From harsh.patel at inextrix.com Wed Nov 1 07:07:11 2023 From: harsh.patel at inextrix.com (Harsh Patel) Date: Wed, 1 Nov 2023 07:07:11 +0000 Subject: [OpenSIPS-Users] Fw: Proxy authorization using MongoDB in Opensips In-Reply-To: References: Message-ID: I am currently using MySQL to proxy authorization using the auth_db module. I am utilizing the proxy_authorize function to authorize users in a MySQL table, and it is working perfectly. However, I need to switch to MongoDB as my database instead of MySQL. The issue is that it appears auth_db only supports MySQL and PostgreSQL databases. My concern is that I must use MongoDB as the database for authorization in OpenSIPS. Is it possible to authorize users in OpenSIPS using MongoDB with the auth_db module, or should I consider an alternative module or approach? Load auth_db module loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://DB_USER:DB_PASSWORD at DB_HOST/DB_NAME") modparam("auth_db", "load_credentials", "$avp(tmp_id)=id") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") check condition in routes. if (!proxy_authorize("", "TABLE)_NAME")) {       proxy_challenge("", "0");       exit; } Best Regards, ---- Harsh Patel Team Lead. Inextrix Technologies Pvt Ltd (www.inextrix.com) Mo: +1 315 898 1049 https://www.inextrix.com/ Disclaimer: The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorized to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance on the contents of this information is strictly prohibited and may be unlawful. Please notify the sender immediately and destroy all copies of this message and any attachments contained in it. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 1 11:12:18 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 1 Nov 2023 13:12:18 +0200 Subject: [OpenSIPS-Users] Fw: Proxy authorization using MongoDB in Opensips In-Reply-To: References: Message-ID: <3a3f58f3-11e0-bea4-35af-570b7a6f221f@opensips.org> Hi, Natively, the auth_db supports only SQL databases. Options you have: 1) try simulating an SQL DB from a noSQL one, by using the db_cachedb module [1] 2) use the pv_proxy_authorize() function [2] and push the credentials from script level - and you can have the prior loaded from Mongo via the cache_fetch() [3] function. [1] https://opensips.org/html/docs/modules/3.4.x/db_cachedb.html [2] https://opensips.org/html/docs/modules/3.4.x/auth.html#func_pv_proxy_authorize [3] https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#cache_fetch Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/1/23 9:07 AM, Harsh Patel via Users wrote: > I am currently using MySQL to proxy authorization using the > *auth_db*module. I am utilizing the *proxy_authorize*function to > authorize users in a MySQL table, and it is working perfectly. > > However, I need to switch to *MongoDB* as my database instead of > MySQL. The issue is that it appears *auth_db* only supports *MySQL* > and *PostgreSQL* databases. *My concern is that I must use MongoDB as > the database for authorization in OpenSIPS. Is it possible to > authorize users in OpenSIPS using MongoDB with the auth_db module, or > should I consider an alternative module or approach?* > * > * > *Load auth_db module* > > loadmodule "auth_db.so" > modparam("auth_db", "db_url", > "mysql://DB_USER:DB_PASSWORD at DB_HOST/DB_NAME") > modparam("auth_db", "load_credentials", "$avp(tmp_id)=id") > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > > > check condition in routes. > > if (!proxy_authorize("", "TABLE)_NAME")) { >       proxy_challenge("", "0"); >       exit; > *}* > > Best Regards, > ---- > Harsh Patel > Team Lead. > Inextrix Technologies Pvt Ltd (www.inextrix.com) > Mo: +1 315 898 1049 > https://www.inextrix.com/ > > Disclaimer: > The information contained in this communication is confidential and > may be legally privileged. It is intended solely for the use of the > individual or entity to whom it is addressed and others authorized to > receive it. If you are not the intended recipient you are hereby > notified that any disclosure, copying, distribution or taking action > in reliance on the contents of this information is strictly prohibited > and may be unlawful. Please notify the sender immediately and destroy > all copies of this message and any attachments contained in it. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Wed Nov 1 14:00:40 2023 From: efes99999 at gmail.com (L S) Date: Wed, 1 Nov 2023 10:00:40 -0400 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: <734f9afa-3f91-4612-9927-880734fba9c7@opensips.org> References: <734f9afa-3f91-4612-9927-880734fba9c7@opensips.org> Message-ID: Thanks Liviu. On Tue, Oct 31, 2023, 9:23 AM Liviu Chircu wrote: > Hi! > > It seems there was an issue related to the tarball packing introduced in > the source tree a couple weeks *before* release day which confused me, as I > thought I was just building the tarball incorrectly on release day (e.g. > maybe due to a dirty directory or a bad script, etc.). > > A fix is now available and all opensips.org tarballs have been rebuilt. > Still, if you were to download the latest stable git tag and run "make tar" > yourself, of course you'd run into the same bug again (*no change there, > the git tag hasn't been moved*)... but that will also get resolved on the > next stable release round in a month or so. > > Best regards, > > On 26.10.2023 09:53, L S wrote: > > Thanks. We are having issues compiling from source on Centos (error > because of a patch related to wolfssl). Will try again. > > > > On Wed, Oct 25, 2023, 4:41 PM Knee Oh via Users > wrote: > >> Yes, compiled from source on Ubuntu 22.04. About to move to production. >> >> >> On Oct 25, 2023, at 4:35 PM, Joseph Jackson >> wrote: >> >>  >> We have but we use the debian packages and we installed it on release day. >> >> >> >> ------------------------------ >> *From:* Users on behalf of L S < >> efes99999 at gmail.com> >> *Sent:* Wednesday, October 25, 2023 12:46 AM >> *To:* OpenSIPS users mailling list >> *Subject:* [OpenSIPS-Users] 3.2.15 installation >> >> Has anyone successfully installed 3.2.15 (revised on Oct 20th)? >> >> Thanks, >> Matt >> >> ------------------------------ >> *From:* Users on behalf of L S < >> efes99999 at gmail.com> >> *Sent:* Wednesday, October 25, 2023 12:46 AM >> *To:* OpenSIPS users mailling list >> *Subject:* [OpenSIPS-Users] 3.2.15 installation >> >> Has anyone successfully installed 3.2.15 (revised on Oct 20th)? >> >> Thanks, >> Matt >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS eBootcamp, Nov 6-17 | www.opensips.org/training > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Thu Nov 2 03:02:22 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Wed, 1 Nov 2023 20:02:22 -0700 Subject: [OpenSIPS-Users] Caller not able to hear Audio file In-Reply-To: References: Message-ID: Devang, Curious if you have found a solution to this problem, as I am having the same problem with Opensips 3.3 and rtpengine. Thank you. Kevin On Mon, Aug 14, 2023, 3:54 AM Devang Dhandhalya via Users < users at lists.opensips.org> wrote: > Hello All > > I am facing the problem that when I make a call, I am not able to hear the > audio file on my softphone(zoiper3&5, microsip) which is playing from > opensips using rtpengine module, but when i capture the that call sip trace > and load that pcap in wireshark i am able to hear audio file music. > > Here is the code snippet: > rtpengine_offer(); > append_to_reply("Content-Type: application/sdp\r\n"); > $var(body) = $(rb{re.subst,/(IP4.).*/\1127.0.0.1/g}); > t_reply_with_body(183, "Session Progress", $var(body)); > rtpengine_play_media("file=/home/file_example_WAV_2MG_G711.org_1.wav"); > > opensips version: 3.2.5 > I am using a Standard wave audio file with 8KHz, 16bit mono, I used other > formats (i.e.mp3) as well but still in softphones I am not able to Hear > voice but in wireshark I am able to hear mp3 file voice. > I would appreciate it if someone has an idea what to do. Please feel free > to ask if you think I have forgotten to describe something that might be > important or something is unclear in what I have written. > > Regards > Devang Dhandhalya > > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Thu Nov 2 08:09:41 2023 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 2 Nov 2023 21:09:41 +1300 Subject: [OpenSIPS-Users] memcached opensips 3.3 Message-ID: Hope everyone is having a good day, I've encountered an issue related to making a memcached group. My understanding is that when configuring a group, OpenSIPS attempts to connect to other memcached servers if one is unresponsive. However, it seems to only connect to the first server in the group and perform insertions exclusively on that one. In the event that the first server becomes inaccessible, I receive an error message stating "Failed to insert: CONNECTION FAILURE" and no connection/insertion to the second one. Here is a snippet of the configuration I'm using: modparam("cachedb_memcached", "cachedb_url","memcached:main://memcacheserver1:11222,memcachedserver2:11222/") I've thoroughly reviewed the documentation available, but I couldn't find sufficient clarity on this behavior. I would greatly appreciate it if someone could provide insights or clarification on this matter. Thanks Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Thu Nov 2 08:11:04 2023 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 2 Nov 2023 21:11:04 +1300 Subject: [OpenSIPS-Users] redis connect with auth In-Reply-To: References: Message-ID: Thank you Bogdan, Without @ it works, the problem is I am unable to change the password since it's being used by a few other apps. Is it possible to escape this? THanks On Fri, 16 Jun 2023 at 01:37, Bogdan-Andrei Iancu wrote: > Hi Jason, > > Most probably the `@` in the pwd is the issue - try to remote it (change > pwd) and see if it works. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 6/15/23 9:14 AM, nz deals wrote: > > Hi everyone, > > I am a newbee and looking for some help. > If i don't set a password for my redis, i can easily connect but when i > set the password i could not connect to redis using opensips. > > My password have @ inside, could this be an issue? > > i am trying like like this - > > modparam("cachedb_redis", "cachedb_url", "redis:main://:mypasx at ds# > ycd at 127.0.0.1:6379/") > > Regards, > Jason > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Thu Nov 2 08:29:56 2023 From: callum.guy at x-on.co.uk (Callum Guy) Date: Thu, 2 Nov 2023 08:29:56 +0000 Subject: [OpenSIPS-Users] Async rest_client timeout In-Reply-To: References: Message-ID: I wanted to follow up with confirmation that opensips is behaving normally here and an issue with delayed HTTP is in fact occurring outside of the server. My apologies for any confusion. Best regards, Callum On Tue, 31 Oct 2023 at 16:21, Callum Guy wrote: > > Hi All, > > I'm seeing a small number of sessions reporting timeouts for async > rest_client post requests. These occur at peak times for system load > and present the following error: > > ERROR:rest_client:_resume_async_http_req: connected, but transfer timed out (5s) > > The return code is -3 and HTTP response arrives as 0. My web server > itself shows that the requests complete in <200ms. The > _resume_async_http_req errors arrive ~400ms after the request was > issued. > > I suspect that something is exhausted (i.e. too many async or curl > sessions) however I am not aware of any configuration options that > would help. > > The following settings are employed: > > modparam("rest_client", "connection_timeout", 4) > modparam("rest_client", "curl_timeout", 5) > modparam("rest_client", "max_async_transfers", 250) > modparam("rest_client", "ssl_verifyhost", 0) > > Can anyone offer advice on where I should be looking to resolve this issue? > > version: opensips 3.2.10 (x86_64/linux) > libcurl-7.76.1-23.el9_2.1.x86_64 > > Many thanks, > > Callum -- Voting link  (it takes less than 20 seconds!) *0333 332 0000  |  x-on.co.uk   |   **      **  |   **Practice Index Reviews * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of X-on Health Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. From bogdan at opensips.org Thu Nov 2 11:23:14 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Nov 2023 13:23:14 +0200 Subject: [OpenSIPS-Users] redis connect with auth In-Reply-To: References: Message-ID: <4bb28c25-b89e-f2b7-6c48-4de18901f942@opensips.org> Hi Jason, right now there is no generic escaping support, so shortly said you cannot use the `@` in the password :(. We can look into this, but please open a Feature Request on the github tracker. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/2/23 10:11 AM, nz deals wrote: > Thank you Bogdan, > Without @ it works, the problem is I am unable to change the password > since it's being used by a few other apps. > Is it possible to escape this? > > THanks > > On Fri, 16 Jun 2023 at 01:37, Bogdan-Andrei Iancu > wrote: > > Hi Jason, > > Most probably the `@` in the pwd is the issue - try to remote it > (change pwd) and see if it works. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 6/15/23 9:14 AM, nz deals wrote: >> Hi everyone, >> >> I am a newbee and looking for some help. >> If i don't set a password for my redis, i can easily connect but >> when i set the password i could not connect to redis using opensips. >> >> My password have @ inside, could this be an issue? >> >> i am trying like like this - >> >> modparam("cachedb_redis", "cachedb_url", >> "redis:main://:mypasx at ds#ycd at 127.0.0.1:6379/ >> ") >> >> Regards, >> Jason >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Nov 2 11:37:46 2023 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 2 Nov 2023 13:37:46 +0200 Subject: [OpenSIPS-Users] redis connect with auth In-Reply-To: <4bb28c25-b89e-f2b7-6c48-4de18901f942@opensips.org> References: <4bb28c25-b89e-f2b7-6c48-4de18901f942@opensips.org> Message-ID: <0f8d8fe8-db9f-60e1-4ab0-18338cdc2be9@opensips.org> On 02.11.2023 13:23, Bogdan-Andrei Iancu wrote: > Hi Jason, > > right now there is no generic escaping support, so shortly said you > cannot use the `@` in the password :(. > > We can look into this, but please open a Feature Request on the github > tracker. Hi guys, I just pushed a quick-fix for this on master branch, see this commit . All previous URL tests are still passing as well as 3 new ones, so I guess we should be good here. Jason, let me know if you can give it a test as well, so I can backport it down to 3.2 stable. Best regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Nov 2 13:34:18 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 2 Nov 2023 19:04:18 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Hi Ben , failure_route[1] { if ( t_check_status("404|477|480|481|408|486|50[234]")){ if (next_branches()) { t_on_branch("attr"); } } } branch_route[attr] { $var(count) = $(hdr(Call-Info){csv.count}); $var(i) = 0; $var(match-count) = 0; while($var(i) < $(var(count))){ if ($(avp(attr){s.index, $(hdr(Call-Info){csv.value,$var(i)})}) != NULL){ xlog("counter: $var(i)th index matched in attribute \n"); $var(match-count)= $var(match-count) + 1; } xlog("counter: $var(i)\n"); $var(i) = $var(i) + 1; } if ($var(i) == $var(match-count)){ ## Here I want to give call to that contact .. if that fails then again it should come to next branch and again compare t_on_failure("1"); route(1); } else{ # Here if the condition does not match . then i want to do the comparison again if (next_branches()){ t_on_branch("attr"); } # drop(); } } As for my expectation, it's not working . How does it work ? Where should I use T_branch_Idx ? Can I get some examples of this ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Oct 27, 2023 at 7:10 PM Ben Newlin wrote: > Just wanted to add that with the solution I recommended below, you would > want to make sure you properly handle the case where all branches were > dropped because none matched. The proper handling in that case for your > system would be defined by your requirements. > > > > Ben Newlin > > > > *From: *Users on behalf of Ben Newlin < > Ben.Newlin at genesys.com> > *Date: *Friday, October 27, 2023 at 9:33 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > ------------------------------ > > Without seeing the specific exact code, I can’t say what is causing that > error. It seems like it may be a syntax issue. For example, in your email > you are missing the semicolon after the line setting the count. I don’t > know if it is that way in your config script. I recommend double checking > all your syntax, and if you still get the error please provide the exact > code from your script for review. > > > > For #2, I can’t really be very specific there because I don’t know all of > your logic, nor am I very knowledgeable about the registrar module. My > first thought is to not do any filtering/checking in the request route. > Just allow the registrar module to load all contacts in branches, but arm a > branch_route. Then perform the check in the branch route for each branch. > If the Contact for the branch doesn’t match what you want, you can drop the > branch [1]. > > > > [1] https://www.opensips.org/Documentation/Script-Routes-3-2#toc2 > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Friday, October 27, 2023 at 6:17 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > ------------------------------ > > Ahhh , Ok . > > Now it's very complicated . When you said a loop that's looping in my > mind . Sorry for the bad joke . > > > > In case , the number of elements in each Invite won't be fixed . So the > number of loops will vary from one Invite to another . > > I was thinking of counting the number of elements first , then as the > index starts from 0 I will loop till *count-1* to fetch every element > properly . > > > > Example : > > $var(count) = $(hdr(Call-Info){csv.count}) ## if the number is 3 then > loop will be for 3 times starting from 0 to 2 > > > > $var(i) = 0; > while($var(i) < $var(count) ) > { > xlog("counter: $var(i)\n"); > $var(i) = $var(i) + 1; > } > > > > This was my initial thought . But while finding the count it gave me an > error . * $var(count) = $(hdr(Call-Info){csv.count}) Is this not in the > correct format ? * > > > > *parse error in /usr/local/etc/opensips/opensips-p2p.cfg:267:26-55: > unknown script variable* > > > > *As I have earlier mentioned my header will look like . * > > *Call-Info: sales,en,level20,en (this can be anything but format will be > like this . ) . How do I count the number of values ?* > > > > *2. As I am doing this matching to filter out contacts, where should I do > this ? If I am doing this while giving a call to the contacts , for the 1st > transaction it is doing the comparison , after that for the next branch it > processes the call without matching . Which is not right . For every > contact this comparison should loop * > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > > > > > > On Thu, Oct 26, 2023 at 7:19 PM Ben Newlin wrote: > > Sasmita, > > > > Apologies, I replied yesterday but the message is being held by the list > as the quoted replies have made it too large. I’ve removed some of the > quoted replies and I’m copying my response below: > > > > Yes, a substring match means the exact complete string exists somewhere in > the string being searched . In your example, the $avp(attr) does not > contain any substring that matches $hdr(Call-Info), so it is correctly > failing. > > > > If you want to check for the presence of each element, you need to loop > through the elements in $hdr(Call-Info) and check for each one in the > $avp(attr) using the s.index mechanism. > > > > The best options for looping on the header are probably s.select [1] or > the csv tranformations [2]. > > > > [1] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 > > [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc82 > > > > Ben Newlin > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Thu Nov 2 19:50:06 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Thu, 2 Nov 2023 12:50:06 -0700 Subject: [OpenSIPS-Users] Caller not able to hear Audio file In-Reply-To: References: Message-ID: I was able to figure out how to do this in Opensips 3.3 rtpengine_manage("from-tag=$ft replace-session-connection trust-address replace-origin codec-strip-g729",,$var(body)); append_to_reply("Contact: \r\n"); append_to_reply("Content-Type: application/sdp\r\n"); t_reply_with_body(183, "SessionProgress", $var(body)); rtpengine_play_media("from-tag=$ft file=/etc/rtpengine/music.wav"); sleep(10); rtpengine_delete("from-tag=$ft"); t_reply(603, "Forbidden") exit; Still having problems with late media invite on this, but trying with 200 OK instead of 183 but not able to absorb the ACK coming back. On Wed, Nov 1, 2023 at 8:02 PM Kevin Kennedy wrote: > Devang, > Curious if you have found a solution to this problem, as I am having the > same problem with Opensips 3.3 and rtpengine. > > Thank you. > > Kevin > > On Mon, Aug 14, 2023, 3:54 AM Devang Dhandhalya via Users < > users at lists.opensips.org> wrote: > >> Hello All >> >> I am facing the problem that when I make a call, I am not able to hear >> the audio file on my softphone(zoiper3&5, microsip) which is playing from >> opensips using rtpengine module, but when i capture the that call sip trace >> and load that pcap in wireshark i am able to hear audio file music. >> >> Here is the code snippet: >> rtpengine_offer(); >> append_to_reply("Content-Type: application/sdp\r\n"); >> $var(body) = $(rb{re.subst,/(IP4.).*/\1127.0.0.1/g}); >> t_reply_with_body(183, "Session Progress", $var(body)); >> >> rtpengine_play_media("file=/home/file_example_WAV_2MG_G711.org_1.wav"); >> >> opensips version: 3.2.5 >> I am using a Standard wave audio file with 8KHz, 16bit mono, I used other >> formats (i.e.mp3) as well but still in softphones I am not able to Hear >> voice but in wireshark I am able to hear mp3 file voice. >> I would appreciate it if someone has an idea what to do. Please feel free >> to ask if you think I have forgotten to describe something that might be >> important or something is unclear in what I have written. >> >> Regards >> Devang Dhandhalya >> >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Thu Nov 2 19:52:53 2023 From: igorolhovskiy at gmail.com (Ihor Olkhovskyi) Date: Thu, 2 Nov 2023 20:52:53 +0100 Subject: [OpenSIPS-Users] OpenSIPS as websocket client Message-ID: Hello, I'm a bit new (to a recent versions) to OpenSIPS and trying it to act as a UDP - WebSocket proxy using it as an outbound proxy in SIP client (PJSUA, if it's important) Currently I'm using 3.4.2 version. Config is quite simple, not far from default one. ... socket=udp:0.0.0.0:6051 socket=wss:0.0.0.0:9443 ... loadmodule "proto_udp.so" loadmodule "proto_tls.so" # WebSocket part loadmodule "proto_wss.so" loadmodule "tls_openssl.so" loadmodule "tls_mgm.so" modparam("tls_mgm", "client_domain", "localhost") modparam("tls_mgm", "certificate", "[localhost]/etc/ssl/certs/ssl-cert-snakeoil.pem") modparam("tls_mgm", "private_key", "[localhost]/etc/ssl/private/ssl-cert-snakeoil.key") modparam("tls_mgm", "ca_list", "[localhost]/etc/ssl/certs/ca-certificates.crt") modparam("tls_mgm", "verify_cert", "[localhost]0") modparam("tls_mgm", "require_cert", "[localhost]0") ... route[relay] { if ($socket_in(proto) == "UDP") { $socket_out = "wss:0.0.0.0:9443"; } else { $socket_out = "udp:0.0.0.0:6051"; } if (!t_relay()) { send_reply(500, "Internal Error"); } exit; } I'm using most generic self-signed certs and just started to make some experiments. But when I'm trying just forward SIP packets to remote server, I'm getting this in the logs DBG:core:parse_headers: flags=ffffffffffffffff DBG:proto_wss:proto_wss_send: no open tcp connection found, opening new one DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 DBG:core:probe_max_sock_buff: using snd buffer of 416 kb DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 4 DBG:core:print_ip: tcpconn_new: new tcp connection to: DBG:core:tcpconn_new: on port 8089, proto 6 DBG:tls_mgm:tls_find_client_domain: found TLS client domain: localhost DBG:tls_openssl:openssl_tls_conn_init: Creating a whole new ssl connection DBG:tls_openssl:openssl_tls_conn_init: Setting in CONNECT mode (client) DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 ERROR:tls_openssl:openssl_tls_blocking_write: TLS send timeout (100) ERROR:proto_wss:ws_client_handshake: cannot start handshake ERROR:proto_wss:ws_connect: cannot complete WebSocket handshake DBG:core:tcpconn_destroy: destroying connection 0x7f0efb106440, flags 0038 DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 NOTICE:tls_openssl:verify_callback: depth = 2, verify success NOTICE:tls_openssl:verify_callback: depth = 1, verify success NOTICE:tls_openssl:verify_callback: depth = 0, verify success INFO:tls_openssl:openssl_tls_connect: New TLS connection to :8089 established DBG:tls_openssl:openssl_tls_connect: new TLS connection to :8089 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 DBG:tls_openssl:openssl_tls_connect: sending socket: 0.0.0.0:37697 INFO:tls_openssl:tls_dump_cert_info: tls_connect: server TLS certificate subject: /CN=*.pbx.company.domain, issuer: /C=GB/ST=Greater Manchester/L=Salford/O=Sectigo Limited/CN=Sectigo RSA Domain Validation Secure Server CA INFO:tls_openssl:tls_dump_cert_info: tls_connect: local TLS client certificate subject: /CN=localhost, issuer: /CN=localhost DBG:tls_openssl:openssl_tls_write: write was successful (6 bytes) DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 DBG:tls_openssl:openssl_tls_write: write was successful (2 bytes) DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 DBG:tls_openssl:openssl_tls_conn_shutdown: first phase of 2-way handshake completed succesfuly ERROR:proto_wss:proto_wss_send: connect failed ERROR:tm:msg_send: send() to :8089 for proto wss/6 failed ERROR:tm:t_forward_nonack: sending request failed DBG:tm:t_relay_to: t_forward_nonack returned error Server that I'm making connections to is supporting TLS and WSS transports. If I'm changing socket type from WSS to TLS, all is working, so it's not a TLS certificate issue or something like this. I'm pretty sure, that I'm missing something obvious, but not really getting what. Would be appreciated for any hints. -- Best regards, Ihor (Igor) -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Nov 2 20:53:14 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 2 Nov 2023 20:53:14 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Sasmita, I can’t provide a working example as I don’t have a use case like this. However, this piece of script you’ve provided does not represent a correct flow. I think you may need to review how the different types of routes, and particularly branch routes, work. [1] I don’t have any experience with Registrar module, so take all of the following with a grain of salt. Someone with more experience with registrar can maybe keep me honest here. You should only need to call next_branches() one time, as it already loads all contacts returned by lookup() into parallel branches (assuming you are using the “b” flag for lookup()). This means they are all sent out at once, not serially. So you don’t need to send the next branch in failure_route because they’ve all already been sent. The branch route is executed as the last route before the message is being sent out. You certainly do not need to call next_branches() there either, in fact its behavior in a branch route is not defined in the docs. Also, I don’t know what your route “1” does, but you likely don’t need it from branch route either. As long as you don’t drop the branch, it will automatically be sent out. Lastly, you have the actual drop() command commented out, so this code won’t work as I described. Lastly, failure_route is armed for the whole request. In the case of parallel branching, it will only be called once for the request, not once for each branch, and only if all branches receive negative replies. One thing I’m not clear about is what happens if you end up dropping all the branches. I don’t know if failure_route would be called then, but it would be pretty easy to verify that. I think it would. Again, I can’t speak to your specific use case, but a representative version of the solution I recommended is below. *I have not tested or verified this code.* route { # all of your normal routing logic if (lookup(“”, “b”)) { if (next_branches()) { t_on_branch(“check_attrs”); t_on_failure(“no_branches”); } else { # handle case of no contacts t_reply(404, “Not Found”); } } else { # handle case of failed lookup t_reply(404, “Not Found”); } } branch_route[check_attrs] { $var(count) = $(hdr(Call-Info){csv.count}); while($(var(count) >= 0)) { if ($(avp(attr){s.index, $(hdr(Call-Info){csv.value,$var(i)})}) == NULL) { # as soon as one requirement doesn’t match, you know you don’t want to route drop(); } xlog("count: $var(count)\n"); $var(count) = $var(count) - 1; } } failure_route[no_branches] { # handle case where all branches failed t_reply(404, “Not Found”); } [1] https://www.opensips.org/Documentation/Script-Routes-3-2 Ben Newlin From: Users on behalf of Sasmita Panda Date: Thursday, November 2, 2023 at 9:36 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . ________________________________ Hi Ben , failure_route[1] { if ( t_check_status("404|477|480|481|408|486|50[234]")){ if (next_branches()) { t_on_branch("attr"); } } } branch_route[attr] { $var(count) = $(hdr(Call-Info){csv.count}); $var(i) = 0; $var(match-count) = 0; while($var(i) < $(var(count))){ if ($(avp(attr){s.index, $(hdr(Call-Info){csv.value,$var(i)})}) != NULL){ xlog("counter: $var(i)th index matched in attribute \n"); $var(match-count)= $var(match-count) + 1; } xlog("counter: $var(i)\n"); $var(i) = $var(i) + 1; } if ($var(i) == $var(match-count)){ ## Here I want to give call to that contact .. if that fails then again it should come to next branch and again compare t_on_failure("1"); route(1); } else{ # Here if the condition does not match . then i want to do the comparison again if (next_branches()){ t_on_branch("attr"); } # drop(); } } As for my expectation, it's not working . How does it work ? Where should I use T_branch_Idx ? Can I get some examples of this ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Thu Nov 2 23:32:02 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Thu, 2 Nov 2023 16:32:02 -0700 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK Message-ID: I am trying to build a solution where Opensips 3.2+ with RTPengine acts as a UAC, answers a call with 200OK, plays media from file, and will terminate the call right after playing announcement. Opensips is responding with 200OK with SDP body and making the correct changes for the IP, but when the ACK comes back from the UAS, Opensips doesn't seem to absorb it and retransmits the 200OK. Code snippet handling this scenario rtpengine_manage("from-tag=$ft replace-session-connection trust-address replace-origin codec-strip-g729",,$var(body)); append_to_reply("Contact: \r\n"); append_to_reply("Content-Type: application/sdp\r\n"); t_reply_with_body(200, "OK", $var(body)); rtpengine_play_media("from-tag=$ft file=/etc/rtpengine/unk_num.wav"); sleep(10); rtpengine_delete("from-tag=$ft"); #t_reply(603, "Decline"); exit(); What do I need to add to handle this scenario correctly? Note: I was able to get this to work with Early Media (183 reply_with_body, and send t_reply(603, "Decline")), but we have customers using late media invite as well, so the Early Media option wouldn't work in that case. Thank you. Kevin Kennedy -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Thu Nov 2 23:44:42 2023 From: nzdealshelp at gmail.com (nz deals) Date: Fri, 3 Nov 2023 12:44:42 +1300 Subject: [OpenSIPS-Users] redis connect with auth In-Reply-To: <0f8d8fe8-db9f-60e1-4ab0-18338cdc2be9@opensips.org> References: <4bb28c25-b89e-f2b7-6c48-4de18901f942@opensips.org> <0f8d8fe8-db9f-60e1-4ab0-18338cdc2be9@opensips.org> Message-ID: Thank you so much Liviu, I can confirm that a password with a special character `@` is working now. Regards, Jason On Fri, 3 Nov 2023 at 00:37, Liviu Chircu wrote: > On 02.11.2023 13:23, Bogdan-Andrei Iancu wrote: > > Hi Jason, > > right now there is no generic escaping support, so shortly said you cannot > use the `@` in the password :(. > > We can look into this, but please open a Feature Request on the github > tracker. > > Hi guys, > > I just pushed a quick-fix for this on master branch, see this commit > > . > > All previous URL tests are still passing as well as 3 new ones, so I guess > we should be good here. Jason, let me know if you can give it a test as > well, so I can backport it down to 3.2 stable. > > Best regards, > > -- > Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Fri Nov 3 00:04:09 2023 From: nzdealshelp at gmail.com (nz deals) Date: Fri, 3 Nov 2023 13:04:09 +1300 Subject: [OpenSIPS-Users] memcached opensips 3.3 In-Reply-To: References: Message-ID: After additional testing, I observed that the settings for Redis are functioning properly, just for your information. It appears there might be an issue specifically when using memcached in a group. Thanks Regards, Jason On Thu, 2 Nov 2023 at 21:09, nz deals wrote: > Hope everyone is having a good day, > I've encountered an issue related to making a memcached group. My > understanding is that when configuring a group, OpenSIPS attempts to > connect to other memcached servers if one is unresponsive. However, it > seems to only connect to the first server in the group and perform > insertions exclusively on that one. In the event that the first server > becomes inaccessible, I receive an error message stating "Failed to insert: > CONNECTION FAILURE" and no connection/insertion to the second one. > > Here is a snippet of the configuration I'm using: > modparam("cachedb_memcached", > "cachedb_url","memcached:main://memcacheserver1:11222,memcachedserver2:11222/") > > I've thoroughly reviewed the documentation available, but I couldn't find > sufficient clarity on this behavior. I would greatly appreciate it if > someone could provide insights or clarification on this matter. > > Thanks > Jason > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Nov 3 10:03:54 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 3 Nov 2023 15:33:54 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: I really appreciate your help Ben . To give some background of current behaviour , 1stly we are not doing parallel forking , we are doing serial forking . We have adjusted opensips code somehow and sorted the list of contacts according to their registration time (*creation time column is added in the location table for this* ). In a branch if there are 5 contacts then we give a call to the longest ideal contact first if that fails then opensips again try the next longest ideal contact and so on . If All fails it gives back 500 Error . My requirements have changed at this point . *Agent1 call-info : sales=1,apple=20,en=5* *Agent2 call-info : sales=15,apple=7* *Agent3 call-info : sales=4,apple=4* *Agent4 Call-info : hr=3,gallileo=5 * *Invite $hdr(Call-Info) : sales,en * *For this INVITE , Agent1 , Agent2 and Agent3 are matched candidates ** and the elements are present in their attribute . Now the question is who should I give the call first . * *Here I have to calculate the average level of both the agent and which one will be the greatest 1st call sent out to that agent . * *When I am saying theoretically it seems achievable through config adjustment . But the question is now how ? * *Agent1 average skill level : (1+20)/2 =10.5* *Agent2 average skill level : (15+7)/2=11 (Now Agent2 is in higher skill . )* *Agent3 average skill level : (4+4)/2=4 * *Now the highest average skill level is Agent2 , Call will be sent out to Agent2 if fails then will go to Agent1 if fails then will go to Agent3 . * *Again this is a challenge for me . Is there any other module or table on which I can save the branch with its skill level and then while sending out call I will check the skill level and then send out . How will I achieve this ?* *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Nov 3, 2023 at 2:28 AM Ben Newlin wrote: > Sasmita, > > > > I can’t provide a working example as I don’t have a use case like this. > However, this piece of script you’ve provided does not represent a correct > flow. I think you may need to review how the different types of routes, and > particularly branch routes, work. [1] > > > > I don’t have any experience with Registrar module, so take all of the > following with a grain of salt. Someone with more experience with registrar > can maybe keep me honest here. > > > > You should only need to call next_branches() one time, as it already loads > all contacts returned by lookup() into parallel branches (assuming you are > using the “b” flag for lookup()). This means they are all sent out at once, > not serially. So you don’t need to send the next branch in failure_route > because they’ve all already been sent. > > > > The branch route is executed as the last route before the message is being > sent out. You certainly do not need to call next_branches() there either, > in fact its behavior in a branch route is not defined in the docs. Also, I > don’t know what your route “1” does, but you likely don’t need it from > branch route either. As long as you don’t drop the branch, it will > automatically be sent out. Lastly, you have the actual drop() command > commented out, so this code won’t work as I described. > > > > Lastly, failure_route is armed for the whole request. In the case of > parallel branching, it will only be called once for the request, not once > for each branch, and only if all branches receive negative replies. > > > > One thing I’m not clear about is what happens if you end up dropping all > the branches. I don’t know if failure_route would be called then, but it > would be pretty easy to verify that. I think it would. > > > > Again, I can’t speak to your specific use case, but a representative > version of the solution I recommended is below. **I have not tested or > verified this code.** > > > > route { > > # all of your normal routing logic > > > > if (lookup(“”, “b”)) { > > if (next_branches()) { > > t_on_branch(“check_attrs”); > > t_on_failure(“no_branches”); > > } > > else { > > # handle case of no contacts > > t_reply(404, “Not Found”); > > } > > } > > else { > > # handle case of failed lookup > > t_reply(404, “Not Found”); > > } > > } > > > branch_route[check_attrs] { > $var(count) = $(hdr(Call-Info){csv.count}); > > while($(var(count) >= 0)) { > if ($(avp(attr){s.index, $(hdr(Call-Info){csv.value,$var(i)})}) == > NULL) { > > # as soon as one requirement doesn’t match, you know you don’t want > to route > > drop(); > > } > > > > xlog("count: $var(count)\n"); > $var(count) = $var(count) - 1; > } > > } > > > > failure_route[no_branches] { > # handle case where all branches failed > > t_reply(404, “Not Found”); > } > > > > [1] https://www.opensips.org/Documentation/Script-Routes-3-2 > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Thursday, November 2, 2023 at 9:36 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > ------------------------------ > > Hi Ben , > > > > > > failure_route[1] { > if ( t_check_status("404|477|480|481|408|486|50[234]")){ > if (next_branches()) > { > t_on_branch("attr"); > } > > } > } > > > branch_route[attr] > { > $var(count) = $(hdr(Call-Info){csv.count}); > > $var(i) = 0; > $var(match-count) = 0; > > while($var(i) < $(var(count))){ > > if ($(avp(attr){s.index, $(hdr(Call-Info){csv.value,$var(i)})}) > != NULL){ > xlog("counter: $var(i)th index matched in attribute \n"); > $var(match-count)= $var(match-count) + 1; > } > xlog("counter: $var(i)\n"); > $var(i) = $var(i) + 1; > } > > if ($var(i) == $var(match-count)){ > ## Here I want to give call to that contact .. if that fails then again it > should come to next branch and again compare > t_on_failure("1"); > route(1); > } > else{ > # Here if the condition does not match . then i want to do the comparison > again > if (next_branches()){ > t_on_branch("attr"); > } > # drop(); > } > } > > > > As for my expectation, it's not working . How does it work ? Where should > I use T_branch_Idx ? Can I get some examples of this ? > > > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From iamhalje at gmail.com Fri Nov 3 11:00:22 2023 From: iamhalje at gmail.com (Dmitry Ponomaryov) Date: Fri, 3 Nov 2023 16:00:22 +0500 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK Message-ID: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Hello everyone, I would like to show my part of the code when playing early media after 200OK, when creating dialogs, I substituted $DLG_did in the contact of my dialog, and received the same $DLG_did for my dialog in ACK, but OpenSIPS also continued to send 200OK , despite having already received an ACK response. route { # initial invite if (is_method("INVITE")) { create_dialog(); route(early_media); exit; } } route[early_media] { if (has_body("application/sdp")) { rtpengine_manage(); } $json(reply) := $rtpquery; $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); remove_body_part(); append_to_reply("Contact: \r\n"); append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); t_reply_with_body(200, "OK", $var(body)); rtpengine_play_media("call-id=$ci from-tag=$ft file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } I don’t know if Kevin example was with creating a dialog, but I also noticed this problem through transaction... thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From iamhalje at gmail.com Fri Nov 3 12:05:45 2023 From: iamhalje at gmail.com (Dmitry Ponomaryov) Date: Fri, 3 Nov 2023 17:05:45 +0500 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: It turns out that this is no early_media, there were simply successful attempts with 183 Session Progress, which is why there was such a misunderstanding, I’ll attach the snippet code again in plain text: route { if (is_method("INVITE")) { create_dialog(); route(media); exit; } } route[media] { if (has_body("application/sdp")) { rtpengine_offer(); } $json(reply) := $rtpquery; $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); remove_body_part(); append_to_reply("Contact: \r\n"); append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); t_reply_with_body(200, "OK", $var(body)); rtpengine_play_media("call-id=$ci from-tag=$ft file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } and pined previous posts below :) > ---------------------------------------------------------------------- > Message: 2 > Date: Fri, 3 Nov 2023 16:00:22 +0500 > From: Dmitry Ponomaryov > To:users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not > absorbing ACK > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> > Content-Type: text/plain; charset="utf-8"; Format="flowed" > > Hello everyone, I would like to show my part of the code when playing > early media after 200OK, when creating dialogs, I substituted $DLG_did > in the contact of my dialog, and received the same $DLG_did for my > dialog in ACK, but OpenSIPS also continued to send 200OK , despite > having already received an ACK response. > > route { > > # initial invite > > if (is_method("INVITE")) { > > create_dialog(); > > route(early_media); > > exit; > > } > > } route[early_media] { if (has_body("application/sdp")) { > rtpengine_manage(); } $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); > remove_body_part(); > > append_to_reply("Contact: > \r\n"); > > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); > t_reply_with_body(200, "OK", $var(body)); > rtpengine_play_media("call-id=$ci from-tag=$ft > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; > } } > > I don’t know if Kevin example was with creating a dialog, but I also > noticed this problem through transaction... thanks > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 2 Nov 2023 16:32:02 -0700 > From: Kevin Kennedy > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not > absorbing ACK > Message-ID: > > Content-Type: text/plain; charset="utf-8" > > I am trying to build a solution where Opensips 3.2+ with RTPengine acts as > a UAC, answers a call with 200OK, plays media from file, and will terminate > the call right after playing announcement. > > Opensips is responding with 200OK with SDP body and making the > correct changes for the IP, but when the ACK comes back from the UAS, > Opensips doesn't seem to absorb it and retransmits the 200OK. > > Code snippet handling this scenario > > rtpengine_manage("from-tag=$ft replace-session-connection > trust-address replace-origin codec-strip-g729",,$var(body)); > append_to_reply("Contact:\r\n"); > append_to_reply("Content-Type: application/sdp\r\n"); > t_reply_with_body(200, "OK", $var(body)); > rtpengine_play_media("from-tag=$ft > file=/etc/rtpengine/unk_num.wav"); > sleep(10); > rtpengine_delete("from-tag=$ft"); > #t_reply(603, "Decline"); > exit(); > > > What do I need to add to handle this scenario correctly? > > Note: I was able to get this to work with Early Media (183 > reply_with_body, and send t_reply(603, "Decline")), but we have customers > using late media invite as well, so the Early Media option wouldn't work in > that case. > > Thank you. > > Kevin Kennedy > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > ------------------------------ > From kennedy4260 at gmail.com Fri Nov 3 19:51:05 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Fri, 3 Nov 2023 12:51:05 -0700 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: Dmitry, Thank you for your response, it does appear to work this way and is absorbing the ACK now, but when a Re-INVITE happens, it responds correctly with the updated Cseq in the 100 Trying, but the 200 OK (using the t_reply_with_body), still has the same Cseq as the initial INVITE. How can I make adjustments for this? Thank you. Kevin On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov wrote: > It turns out that this is no early_media, there were simply successful > attempts with 183 Session Progress, which is why there was such a > misunderstanding, I’ll attach the snippet code again in plain text: > route { if (is_method("INVITE")) { create_dialog(); route(media); exit; > } } route[media] { if (has_body("application/sdp")) { rtpengine_offer(); > } $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); > remove_body_part(); append_to_reply("Contact: > \r\n"); > > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); > t_reply_with_body(200, "OK", $var(body)); > rtpengine_play_media("call-id=$ci from-tag=$ft > file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } > route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; > } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } > > and pined previous posts below :) > > > ---------------------------------------------------------------------- > > Message: 2 > > Date: Fri, 3 Nov 2023 16:00:22 +0500 > > From: Dmitry Ponomaryov > > To:users at lists.opensips.org > > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not > > absorbing ACK > > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> > > Content-Type: text/plain; charset="utf-8"; Format="flowed" > > > > Hello everyone, I would like to show my part of the code when playing > > early media after 200OK, when creating dialogs, I substituted $DLG_did > > in the contact of my dialog, and received the same $DLG_did for my > > dialog in ACK, but OpenSIPS also continued to send 200OK , despite > > having already received an ACK response. > > > > route { > > > > # initial invite > > > > if (is_method("INVITE")) { > > > > create_dialog(); > > > > route(early_media); > > > > exit; > > > > } > > > > } route[early_media] { if (has_body("application/sdp")) { > > rtpengine_manage(); } $json(reply) := $rtpquery; > > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); > > remove_body_part(); > > > > append_to_reply("Contact: > > $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n"); > > > > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = > > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = > > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); > > t_reply_with_body(200, "OK", $var(body)); > > rtpengine_play_media("call-id=$ci from-tag=$ft > > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } > > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); > > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; > > } } > > > > I don’t know if Kevin example was with creating a dialog, but I also > > noticed this problem through transaction... thanks > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL:< > http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html > > > > ---------------------------------------------------------------------- > > > > Message: 1 > > Date: Thu, 2 Nov 2023 16:32:02 -0700 > > From: Kevin Kennedy > > To: OpenSIPS users mailling list > > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not > > absorbing ACK > > Message-ID: > > < > CABDXsRxLTp2_uEX_UPX1adg16af6gaetzJujUTPki8c7H3KKLQ at mail.gmail.com> > > Content-Type: text/plain; charset="utf-8" > > > > I am trying to build a solution where Opensips 3.2+ with RTPengine acts > as > > a UAC, answers a call with 200OK, plays media from file, and will > terminate > > the call right after playing announcement. > > > > Opensips is responding with 200OK with SDP body and making the > > correct changes for the IP, but when the ACK comes back from the UAS, > > Opensips doesn't seem to absorb it and retransmits the 200OK. > > > > Code snippet handling this scenario > > > > rtpengine_manage("from-tag=$ft replace-session-connection > > trust-address replace-origin codec-strip-g729",,$var(body)); > > append_to_reply("Contact:\r\n"); > > append_to_reply("Content-Type: application/sdp\r\n"); > > t_reply_with_body(200, "OK", $var(body)); > > rtpengine_play_media("from-tag=$ft > > file=/etc/rtpengine/unk_num.wav"); > > sleep(10); > > rtpengine_delete("from-tag=$ft"); > > #t_reply(603, "Decline"); > > exit(); > > > > > > What do I need to add to handle this scenario correctly? > > > > Note: I was able to get this to work with Early Media (183 > > reply_with_body, and send t_reply(603, "Decline")), but we have customers > > using late media invite as well, so the Early Media option wouldn't work > in > > that case. > > > > Thank you. > > > > Kevin Kennedy > > -------------- next part -------------- > > An HTML attachment was scrubbed... > > URL:< > http://lists.opensips.org/pipermail/users/attachments/20231102/dd52d307/attachment-0001.html > > > > > > ------------------------------ > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Sun Nov 5 11:35:39 2023 From: igorolhovskiy at gmail.com (Ihor Olkhovskyi) Date: Sun, 5 Nov 2023 12:35:39 +0100 Subject: [OpenSIPS-Users] OpenSIPS as websocket client In-Reply-To: References: Message-ID: Seems to be, default timeouts are too low. By adding modparam("proto_wss", "wss_handshake_timeout", 500) modparam("proto_wss", "wss_tls_handshake_timeout", 500) everything is working. Le 02/11/2023 à 20:52, Ihor Olkhovskyi a écrit : > Hello, > I'm a bit new (to a recent versions) to OpenSIPS and trying it to act > as a UDP - WebSocket proxy using it as an outbound proxy in SIP client > (PJSUA, if it's important) > > Currently I'm using 3.4.2 version. > Config is quite simple, not far from default one. > ... > socket=udp:0.0.0.0:6051 > socket=wss:0.0.0.0:9443 > ... > loadmodule "proto_udp.so" > loadmodule "proto_tls.so" > > # WebSocket part > loadmodule "proto_wss.so" > > loadmodule "tls_openssl.so" > loadmodule "tls_mgm.so" > > modparam("tls_mgm", "client_domain", "localhost") > modparam("tls_mgm", "certificate", > "[localhost]/etc/ssl/certs/ssl-cert-snakeoil.pem") > modparam("tls_mgm", "private_key", > "[localhost]/etc/ssl/private/ssl-cert-snakeoil.key") > modparam("tls_mgm", "ca_list", > "[localhost]/etc/ssl/certs/ca-certificates.crt") > modparam("tls_mgm", "verify_cert", "[localhost]0") > modparam("tls_mgm", "require_cert", "[localhost]0") > > ... > route[relay] { >     if ($socket_in(proto) == "UDP") { >         $socket_out = "wss:0.0.0.0:9443 "; >     } else { >         $socket_out = "udp:0.0.0.0:6051 "; >     } > >     if (!t_relay()) { >         send_reply(500, "Internal Error"); >     } >     exit; > } > > I'm using most generic self-signed certs and just started to make some > experiments. > But when I'm trying just forward SIP packets to remote server, I'm > getting this in the logs > > DBG:core:parse_headers: flags=ffffffffffffffff > DBG:proto_wss:proto_wss_send: no open tcp connection found, opening > new one > DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384 > DBG:core:probe_max_sock_buff: using snd buffer of 416 kb > DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 4 > DBG:core:print_ip: tcpconn_new: new tcp connection to: > DBG:core:tcpconn_new: on port 8089, proto 6 > DBG:tls_mgm:tls_find_client_domain: found TLS client domain: localhost > DBG:tls_openssl:openssl_tls_conn_init: Creating a whole new ssl connection > DBG:tls_openssl:openssl_tls_conn_init: Setting in CONNECT mode (client) > DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 > ERROR:tls_openssl:openssl_tls_blocking_write: TLS send timeout (100) > ERROR:proto_wss:ws_client_handshake: cannot start handshake > ERROR:proto_wss:ws_connect: cannot complete WebSocket handshake > DBG:core:tcpconn_destroy: destroying connection 0x7f0efb106440, flags 0038 > DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 > NOTICE:tls_openssl:verify_callback: depth = 2, verify success > NOTICE:tls_openssl:verify_callback: depth = 1, verify success > NOTICE:tls_openssl:verify_callback: depth = 0, verify success > INFO:tls_openssl:openssl_tls_connect: New TLS connection to > :8089 established > DBG:tls_openssl:openssl_tls_connect: new TLS connection to > :8089 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 > DBG:tls_openssl:openssl_tls_connect: sending socket: 0.0.0.0:37697 > > INFO:tls_openssl:tls_dump_cert_info: tls_connect: server TLS > certificate subject: /CN=*.pbx.company.domain, issuer: > /C=GB/ST=Greater Manchester/L=Salford/O=Sectigo Limited/CN=Sectigo RSA > Domain Validation Secure Server CA > INFO:tls_openssl:tls_dump_cert_info: tls_connect: local TLS client > certificate subject: /CN=localhost, issuer: /CN=localhost > DBG:tls_openssl:openssl_tls_write: write was successful (6 bytes) > DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 > DBG:tls_openssl:openssl_tls_write: write was successful (2 bytes) > DBG:tls_openssl:openssl_tls_update_fd: New fd is 4 > DBG:tls_openssl:openssl_tls_conn_shutdown: first phase of 2-way > handshake completed succesfuly > ERROR:proto_wss:proto_wss_send: connect failed > ERROR:tm:msg_send: send() to :8089 for proto wss/6 failed > ERROR:tm:t_forward_nonack: sending request failed > DBG:tm:t_relay_to: t_forward_nonack returned error > > > Server that I'm making connections to is supporting TLS and WSS > transports. If I'm changing socket type from WSS to TLS, all is > working, so it's not a TLS certificate issue or something like this. > > I'm pretty sure, that I'm missing something obvious, but not really > getting what. > > Would be appreciated for any hints. > -- > Best regards, > Ihor (Igor) -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Mon Nov 6 20:54:11 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Mon, 6 Nov 2023 12:54:11 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there was some re-invite fixes. Still doesn't seem to resolve this issue. What am I missing to handle this correctly? Thank you. Kevin On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy wrote: > Dmitry, > Thank you for your response, it does appear to work this way and is > absorbing the ACK now, but when a Re-INVITE happens, it responds correctly > with the updated Cseq in the 100 Trying, but the 200 OK (using the > t_reply_with_body), still has the same Cseq as the initial INVITE. How can > I make adjustments for this? > > Thank you. > > Kevin > > On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov > wrote: > >> It turns out that this is no early_media, there were simply successful >> attempts with 183 Session Progress, which is why there was such a >> misunderstanding, I’ll attach the snippet code again in plain text: >> route { if (is_method("INVITE")) { create_dialog(); route(media); exit; >> } } route[media] { if (has_body("application/sdp")) { rtpengine_offer(); >> } $json(reply) := $rtpquery; >> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >> remove_body_part(); append_to_reply("Contact: >> \r\n"); >> >> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >> t_reply_with_body(200, "OK", $var(body)); >> rtpengine_play_media("call-id=$ci from-tag=$ft >> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } >> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; >> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } >> >> and pined previous posts below :) >> >> > ---------------------------------------------------------------------- >> > Message: 2 >> > Date: Fri, 3 Nov 2023 16:00:22 +0500 >> > From: Dmitry Ponomaryov >> > To:users at lists.opensips.org >> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not >> > absorbing ACK >> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> >> > Content-Type: text/plain; charset="utf-8"; Format="flowed" >> > >> > Hello everyone, I would like to show my part of the code when playing >> > early media after 200OK, when creating dialogs, I substituted $DLG_did >> > in the contact of my dialog, and received the same $DLG_did for my >> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite >> > having already received an ACK response. >> > >> > route { >> > >> > # initial invite >> > >> > if (is_method("INVITE")) { >> > >> > create_dialog(); >> > >> > route(early_media); >> > >> > exit; >> > >> > } >> > >> > } route[early_media] { if (has_body("application/sdp")) { >> > rtpengine_manage(); } $json(reply) := $rtpquery; >> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >> > remove_body_part(); >> > >> > append_to_reply("Contact: >> > > $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n"); >> > >> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >> > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >> > t_reply_with_body(200, "OK", $var(body)); >> > rtpengine_play_media("call-id=$ci from-tag=$ft >> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } >> > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); >> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; >> > } } >> > >> > I don’t know if Kevin example was with creating a dialog, but I also >> > noticed this problem through transaction... thanks >> > -------------- next part -------------- >> > An HTML attachment was scrubbed... >> > URL:< >> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html >> > >> > ---------------------------------------------------------------------- >> > >> > Message: 1 >> > Date: Thu, 2 Nov 2023 16:32:02 -0700 >> > From: Kevin Kennedy >> > To: OpenSIPS users mailling list >> > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not >> > absorbing ACK >> > Message-ID: >> > < >> CABDXsRxLTp2_uEX_UPX1adg16af6gaetzJujUTPki8c7H3KKLQ at mail.gmail.com> >> > Content-Type: text/plain; charset="utf-8" >> > >> > I am trying to build a solution where Opensips 3.2+ with RTPengine acts >> as >> > a UAC, answers a call with 200OK, plays media from file, and will >> terminate >> > the call right after playing announcement. >> > >> > Opensips is responding with 200OK with SDP body and making the >> > correct changes for the IP, but when the ACK comes back from the UAS, >> > Opensips doesn't seem to absorb it and retransmits the 200OK. >> > >> > Code snippet handling this scenario >> > >> > rtpengine_manage("from-tag=$ft replace-session-connection >> > trust-address replace-origin codec-strip-g729",,$var(body)); >> > append_to_reply("Contact:\r\n"); >> > append_to_reply("Content-Type: application/sdp\r\n"); >> > t_reply_with_body(200, "OK", $var(body)); >> > rtpengine_play_media("from-tag=$ft >> > file=/etc/rtpengine/unk_num.wav"); >> > sleep(10); >> > rtpengine_delete("from-tag=$ft"); >> > #t_reply(603, "Decline"); >> > exit(); >> > >> > >> > What do I need to add to handle this scenario correctly? >> > >> > Note: I was able to get this to work with Early Media (183 >> > reply_with_body, and send t_reply(603, "Decline")), but we have >> customers >> > using late media invite as well, so the Early Media option wouldn't >> work in >> > that case. >> > >> > Thank you. >> > >> > Kevin Kennedy >> > -------------- next part -------------- >> > An HTML attachment was scrubbed... >> > URL:< >> http://lists.opensips.org/pipermail/users/attachments/20231102/dd52d307/attachment-0001.html >> > >> > >> > ------------------------------ >> > >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Mon Nov 6 22:58:57 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Mon, 6 Nov 2023 14:58:57 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: I would like to clarify the issue in case its not 100% clear. * Caller sends INVITE with No SDP(Late Media Invite) * Another device in path (B2BUA) receives INVITE and sends dummy SDP to Opensips with just G.711 codec in the Offer * Opensips Creates Dialog and sends 200OK with SDP using t_reply_with_body based on previously provided information. * B2BUA receives 200OK with SDP then sends ACK followed by a Re-INVITE with No SDP back to OpenSips. * Opensips appears to accept the ACK as it doesn't retransmit the 200OK right away as before updated changes. * Opensips sends 100 trying with new CSEQ from Re-INVITE with no SDP * 200OK Loop created * Opensips send 200 OK with old CSEQ * B2BUA sends ACK with old CSEQ * Call times out. No audio sent Thank you Kevin. * On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy wrote: > I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there > was some re-invite fixes. Still doesn't seem to resolve this issue. What > am I missing to handle this correctly? > > Thank you. > > Kevin > > On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy > wrote: > >> Dmitry, >> Thank you for your response, it does appear to work this way and is >> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly >> with the updated Cseq in the 100 Trying, but the 200 OK (using the >> t_reply_with_body), still has the same Cseq as the initial INVITE. How can >> I make adjustments for this? >> >> Thank you. >> >> Kevin >> >> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov >> wrote: >> >>> It turns out that this is no early_media, there were simply successful >>> attempts with 183 Session Progress, which is why there was such a >>> misunderstanding, I’ll attach the snippet code again in plain text: >>> route { if (is_method("INVITE")) { create_dialog(); route(media); exit; >>> } } route[media] { if (has_body("application/sdp")) { rtpengine_offer(); >>> } $json(reply) := $rtpquery; >>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>> remove_body_part(); append_to_reply("Contact: >>> \r\n"); >>> >>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>> t_reply_with_body(200, "OK", $var(body)); >>> rtpengine_play_media("call-id=$ci from-tag=$ft >>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } >>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; >>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } >>> >>> and pined previous posts below :) >>> >>> > ---------------------------------------------------------------------- >>> > Message: 2 >>> > Date: Fri, 3 Nov 2023 16:00:22 +0500 >>> > From: Dmitry Ponomaryov >>> > To:users at lists.opensips.org >>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>> > absorbing ACK >>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> >>> > Content-Type: text/plain; charset="utf-8"; Format="flowed" >>> > >>> > Hello everyone, I would like to show my part of the code when playing >>> > early media after 200OK, when creating dialogs, I substituted $DLG_did >>> > in the contact of my dialog, and received the same $DLG_did for my >>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite >>> > having already received an ACK response. >>> > >>> > route { >>> > >>> > # initial invite >>> > >>> > if (is_method("INVITE")) { >>> > >>> > create_dialog(); >>> > >>> > route(early_media); >>> > >>> > exit; >>> > >>> > } >>> > >>> > } route[early_media] { if (has_body("application/sdp")) { >>> > rtpengine_manage(); } $json(reply) := $rtpquery; >>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local >>> port); >>> > remove_body_part(); >>> > >>> > append_to_reply("Contact: >>> > >> $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n"); >>> > >>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>> > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>> > t_reply_with_body(200, "OK", $var(body)); >>> > rtpengine_play_media("call-id=$ci from-tag=$ft >>> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } >>> > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); >>> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); >>> exit; >>> > } } >>> > >>> > I don’t know if Kevin example was with creating a dialog, but I also >>> > noticed this problem through transaction... thanks >>> > -------------- next part -------------- >>> > An HTML attachment was scrubbed... >>> > URL:< >>> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html >>> > >>> > ---------------------------------------------------------------------- >>> > >>> > Message: 1 >>> > Date: Thu, 2 Nov 2023 16:32:02 -0700 >>> > From: Kevin Kennedy >>> > To: OpenSIPS users mailling list >>> > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>> > absorbing ACK >>> > Message-ID: >>> > < >>> CABDXsRxLTp2_uEX_UPX1adg16af6gaetzJujUTPki8c7H3KKLQ at mail.gmail.com> >>> > Content-Type: text/plain; charset="utf-8" >>> > >>> > I am trying to build a solution where Opensips 3.2+ with RTPengine >>> acts as >>> > a UAC, answers a call with 200OK, plays media from file, and will >>> terminate >>> > the call right after playing announcement. >>> > >>> > Opensips is responding with 200OK with SDP body and making the >>> > correct changes for the IP, but when the ACK comes back from the UAS, >>> > Opensips doesn't seem to absorb it and retransmits the 200OK. >>> > >>> > Code snippet handling this scenario >>> > >>> > rtpengine_manage("from-tag=$ft replace-session-connection >>> > trust-address replace-origin codec-strip-g729",,$var(body)); >>> > append_to_reply("Contact:\r\n"); >>> > append_to_reply("Content-Type: application/sdp\r\n"); >>> > t_reply_with_body(200, "OK", $var(body)); >>> > rtpengine_play_media("from-tag=$ft >>> > file=/etc/rtpengine/unk_num.wav"); >>> > sleep(10); >>> > rtpengine_delete("from-tag=$ft"); >>> > #t_reply(603, "Decline"); >>> > exit(); >>> > >>> > >>> > What do I need to add to handle this scenario correctly? >>> > >>> > Note: I was able to get this to work with Early Media (183 >>> > reply_with_body, and send t_reply(603, "Decline")), but we have >>> customers >>> > using late media invite as well, so the Early Media option wouldn't >>> work in >>> > that case. >>> > >>> > Thank you. >>> > >>> > Kevin Kennedy >>> > -------------- next part -------------- >>> > An HTML attachment was scrubbed... >>> > URL:< >>> http://lists.opensips.org/pipermail/users/attachments/20231102/dd52d307/attachment-0001.html >>> > >>> > >>> > ------------------------------ >>> > >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue Nov 7 07:00:00 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 7 Nov 2023 12:30:00 +0530 Subject: [OpenSIPS-Users] Need some help in lookup flag "B" in opensips 3.2 version . Message-ID: Hi All , I am using opensips 3.2 version . route { ------- if ($rm=="INVITE") { if(!lookup("location","B")) { if (!t_reply(404, "Not Found")) { sl_reply_error(); } exit; } } if (!serialize_branches(1)){ sl_send_reply(500,"Unable to load contacts"); exit; }else{ # if (next_branches()){ t_on_failure("1"); } } Then of course the route(1) and failure_route(1) i have called . But what is happening in my case is . I have 2 branches , For Invite opensips tries both the branches but if both branch wont accept the call Opensips should reply error code to the caller . But Opensips create a new invite as the Caller sent and again follows the lookup logic . Why is this happening ? How will I manage this ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 7 11:49:13 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Nov 2023 13:49:13 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Call for Location Message-ID: <4d4c8874-00d3-194a-eb31-59df6c70d64d@opensips.org> Hello, It is time to start planning the 2024 OpenSIPS Summit. And first thing to do is to pick up a location. At this stage we are looking for nominations, for cities (in Europe) and local help/support. We need all the help we can get in order to put together yet another great edition of this event 😉. https://blog.opensips.org/2023/11/07/opensips-summit-2024-call-for-location/ Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com From Johan at democon.be Tue Nov 7 11:55:47 2023 From: Johan at democon.be (Johan De Clercq) Date: Tue, 7 Nov 2023 12:55:47 +0100 Subject: [OpenSIPS-Users] opensips summit Message-ID: list, do we have somebody who can arrange something in Vienna ? looks like a great location to me :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: From sh at h-m.net Tue Nov 7 14:06:04 2023 From: sh at h-m.net (Stefan Hofmeir) Date: Tue, 7 Nov 2023 15:06:04 +0100 Subject: [OpenSIPS-Users] opensips summit In-Reply-To: References: Message-ID: <12710221473.20231107150604@h-m.net> Hi, perhaps Munich would be also a great city for the next OpenSIPS Summit. As a native of Munich I could organize recommending hotels or the social event. --  BRs Stefan From Johan at democon.be Tue Nov 7 14:14:44 2023 From: Johan at democon.be (Johan De Clercq) Date: Tue, 7 Nov 2023 15:14:44 +0100 Subject: [OpenSIPS-Users] opensips summit In-Reply-To: <12710221473.20231107150604@h-m.net> References: <12710221473.20231107150604@h-m.net> Message-ID: good plan. I haven't visited munich in ages. Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir : > Hi, > > perhaps Munich would be also a great city for the next OpenSIPS Summit. > As a native of Munich I could organize recommending hotels or the social > event. > > -- > BRs > Stefan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Tue Nov 7 17:19:38 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Tue, 7 Nov 2023 09:19:38 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: Looks like in the debug it is not finding an SDP so it cant construct a response. Is there a way to remedy this? Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:rtpengine:extract_body: No body found Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] ERROR:rtpengine:rtpengine_offer_answer_body: can't extract body from the message Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=40 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_to_param: tag=1985761840-1699375436084- Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_to_param: end of header reached, state=11 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:_parse_to: end of header reached, state=29 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:_parse_to: display={}, ruri={sip:+17024054893 at 10.20.252.101 ;user=phone} Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:rtpengine:rtpe_function_call: proxy reply: d7:createdi1699375436e10:created_usi125586e11:last signali1699375436e4:SSRCde4:tagsde6:totalsd3:RTPd7:packetsi0e5:bytesi0e6:errorsi0ee4:RTCPd7:packetsi0e5:bytesi0e6:errorsi0eee6:result2:oke Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=ffffffffffffffff Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:sipmsgops:remove_body_part_f: no body found Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=ffffffffffffffff Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:pv_get_msg_body: no message body Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] ERROR:core:get_cmd_fixups: Variable in param [3] is not a string Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] ERROR:core:do_action: Failed to get fixups for command in /etc/opensips/opensips.cfg, line 284 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:rtpengine:rtpe_function_call: proxy reply: d8:durationi10152e6:result2:oke Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:t_newtran: transaction on entrance=(nil) Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=ffffffffffffffff Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=78 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:t_lookup_request: start searching: hash=31608, isACK=0 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:matching_3261: RFC3261 transaction matching failed Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:t_lookup_request: no transaction found Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:run_any_trans_callbacks: trans=0x7f920581abe8, callback type 1, id 0 entered Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:core:parse_headers: flags=ffffffffffffffff Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:_reply_light: reply sent out. buf=0x7f92094d8218: SIP/2.0 1..., shmem=0x7f920581dfa8: SIP/2.0 1 Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] DBG:tm:_reply_light: finished Thank you. Kevin On Mon, Nov 6, 2023 at 2:58 PM Kevin Kennedy wrote: > I would like to clarify the issue in case its not 100% clear. > * Caller sends INVITE with No SDP(Late Media Invite) > * Another device in path (B2BUA) receives INVITE and sends dummy SDP to > Opensips with just G.711 codec in the Offer > * Opensips Creates Dialog and sends 200OK with SDP using t_reply_with_body > based on previously provided information. > * B2BUA receives 200OK with SDP then sends ACK followed by a Re-INVITE > with No SDP back to OpenSips. > * Opensips appears to accept the ACK as it doesn't retransmit the 200OK > right away as before updated changes. > * Opensips sends 100 trying with new CSEQ from Re-INVITE with no SDP > * 200OK Loop created > * Opensips send 200 OK with old CSEQ > * B2BUA sends ACK with old CSEQ > * Call times out. > > No audio sent > > Thank you > > Kevin. > * > > > > On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy > wrote: > >> I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there >> was some re-invite fixes. Still doesn't seem to resolve this issue. What >> am I missing to handle this correctly? >> >> Thank you. >> >> Kevin >> >> On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy >> wrote: >> >>> Dmitry, >>> Thank you for your response, it does appear to work this way and is >>> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly >>> with the updated Cseq in the 100 Trying, but the 200 OK (using the >>> t_reply_with_body), still has the same Cseq as the initial INVITE. How can >>> I make adjustments for this? >>> >>> Thank you. >>> >>> Kevin >>> >>> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov >>> wrote: >>> >>>> It turns out that this is no early_media, there were simply successful >>>> attempts with 183 Session Progress, which is why there was such a >>>> misunderstanding, I’ll attach the snippet code again in plain text: >>>> route { if (is_method("INVITE")) { create_dialog(); route(media); exit; >>>> } } route[media] { if (has_body("application/sdp")) { >>>> rtpengine_offer(); >>>> } $json(reply) := $rtpquery; >>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local >>>> port); >>>> remove_body_part(); append_to_reply("Contact: >>>> \r\n"); >>>> >>>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>> t_reply_with_body(200, "OK", $var(body)); >>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } >>>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; >>>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } >>>> >>>> and pined previous posts below :) >>>> >>>> > ---------------------------------------------------------------------- >>>> > Message: 2 >>>> > Date: Fri, 3 Nov 2023 16:00:22 +0500 >>>> > From: Dmitry Ponomaryov >>>> > To:users at lists.opensips.org >>>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>>> > absorbing ACK >>>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> >>>> > Content-Type: text/plain; charset="utf-8"; Format="flowed" >>>> > >>>> > Hello everyone, I would like to show my part of the code when playing >>>> > early media after 200OK, when creating dialogs, I substituted $DLG_did >>>> > in the contact of my dialog, and received the same $DLG_did for my >>>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite >>>> > having already received an ACK response. >>>> > >>>> > route { >>>> > >>>> > # initial invite >>>> > >>>> > if (is_method("INVITE")) { >>>> > >>>> > create_dialog(); >>>> > >>>> > route(early_media); >>>> > >>>> > exit; >>>> > >>>> > } >>>> > >>>> > } route[early_media] { if (has_body("application/sdp")) { >>>> > rtpengine_manage(); } $json(reply) := $rtpquery; >>>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local >>>> port); >>>> > remove_body_part(); >>>> > >>>> > append_to_reply("Contact: >>>> > >>> $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n"); >>>> > >>>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>>> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>>> > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>> > t_reply_with_body(200, "OK", $var(body)); >>>> > rtpengine_play_media("call-id=$ci from-tag=$ft >>>> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); >>>> } >>>> > route[after_early_media] { if (t_was_cancelled()) { >>>> rtpengine_delete(); >>>> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); >>>> exit; >>>> > } } >>>> > >>>> > I don’t know if Kevin example was with creating a dialog, but I also >>>> > noticed this problem through transaction... thanks >>>> > -------------- next part -------------- >>>> > An HTML attachment was scrubbed... >>>> > URL:< >>>> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html >>>> > >>>> > ---------------------------------------------------------------------- >>>> > >>>> > Message: 1 >>>> > Date: Thu, 2 Nov 2023 16:32:02 -0700 >>>> > From: Kevin Kennedy >>>> > To: OpenSIPS users mailling list >>>> > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>>> > absorbing ACK >>>> > Message-ID: >>>> > < >>>> CABDXsRxLTp2_uEX_UPX1adg16af6gaetzJujUTPki8c7H3KKLQ at mail.gmail.com> >>>> > Content-Type: text/plain; charset="utf-8" >>>> > >>>> > I am trying to build a solution where Opensips 3.2+ with RTPengine >>>> acts as >>>> > a UAC, answers a call with 200OK, plays media from file, and will >>>> terminate >>>> > the call right after playing announcement. >>>> > >>>> > Opensips is responding with 200OK with SDP body and making the >>>> > correct changes for the IP, but when the ACK comes back from the UAS, >>>> > Opensips doesn't seem to absorb it and retransmits the 200OK. >>>> > >>>> > Code snippet handling this scenario >>>> > >>>> > rtpengine_manage("from-tag=$ft replace-session-connection >>>> > trust-address replace-origin codec-strip-g729",,$var(body)); >>>> > append_to_reply("Contact:\r\n"); >>>> > append_to_reply("Content-Type: application/sdp\r\n"); >>>> > t_reply_with_body(200, "OK", $var(body)); >>>> > rtpengine_play_media("from-tag=$ft >>>> > file=/etc/rtpengine/unk_num.wav"); >>>> > sleep(10); >>>> > rtpengine_delete("from-tag=$ft"); >>>> > #t_reply(603, "Decline"); >>>> > exit(); >>>> > >>>> > >>>> > What do I need to add to handle this scenario correctly? >>>> > >>>> > Note: I was able to get this to work with Early Media (183 >>>> > reply_with_body, and send t_reply(603, "Decline")), but we have >>>> customers >>>> > using late media invite as well, so the Early Media option wouldn't >>>> work in >>>> > that case. >>>> > >>>> > Thank you. >>>> > >>>> > Kevin Kennedy >>>> > -------------- next part -------------- >>>> > An HTML attachment was scrubbed... >>>> > URL:< >>>> http://lists.opensips.org/pipermail/users/attachments/20231102/dd52d307/attachment-0001.html >>>> > >>>> > >>>> > ------------------------------ >>>> > >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Wed Nov 8 10:50:37 2023 From: igorolhovskiy at gmail.com (Ihor Olkhovskyi) Date: Wed, 8 Nov 2023 11:50:37 +0100 Subject: [OpenSIPS-Users] opensips summit In-Reply-To: References: <12710221473.20231107150604@h-m.net> Message-ID: I can propose Geneva, but I'm afraid people will be chocked with local prices. But can organise a visit to CERN Le mar. 7 nov. 2023 à 15:19, Johan De Clercq a écrit : > good plan. I haven't visited munich in ages. > > Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir : > >> Hi, >> >> perhaps Munich would be also a great city for the next OpenSIPS Summit. >> As a native of Munich I could organize recommending hotels or the social >> event. >> >> -- >> BRs >> Stefan >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Best regards, Ihor (Igor) -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Nov 8 13:08:27 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 8 Nov 2023 18:38:27 +0530 Subject: [OpenSIPS-Users] Need some help in drop() core function of opensips . Message-ID: Hi All , branch_route[1] { xlog("L_NOTICE", " Branch route URI : $(branch(uri)) branch index : $T_branch_idx \n"); xlog(" current branch Q value : $(branch(q)) "); if (isbflagset("Invalid")){ xlog(" dropping current branch : $(branch(uri)[$T_branch_idx]) "); drop; } } 1. When drop() is called , it sends *"500 Server error occurred" . I want a custom error response on this . Is this possible ? If yes , then how will I do this ? * *2.* Inside the branch route I wanted to print the* branch uri and q value* , but it's giving me NULL . Am I doing something wrong here ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 8 13:47:52 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 8 Nov 2023 15:47:52 +0200 Subject: [OpenSIPS-Users] opensips summit In-Reply-To: References: <12710221473.20231107150604@h-m.net> Message-ID: <3ddc2330-90af-5e5b-39be-30cd23d40c11@opensips.org> Hi Ihor, If you can provide local assistance in Geneva, maybe you should consider submitting this by email...and let's see how it's comparing with the other options we get :) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/8/23 12:50 PM, Ihor Olkhovskyi wrote: > I can propose Geneva, but I'm afraid people will be chocked with local > prices. > But can organise a visit to CERN > > Le mar. 7 nov. 2023 à 15:19, Johan De Clercq > a écrit : > > good plan.  I haven't visited munich in ages. > > Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir >: > > Hi, > > perhaps Munich would be also a great city for the next > OpenSIPS Summit. > As a native of Munich I could organize recommending hotels or > the social event. > > -- > BRs > Stefan > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Best regards, > Ihor (Igor) > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From akogan at 5gfuture.com Thu Nov 9 11:07:34 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Thu, 9 Nov 2023 15:07:34 +0400 Subject: [OpenSIPS-Users] Not enough free pkg memory Message-ID: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> Hello all, I regularly get a memory error after upgrading to 3.2.14 and 3.2.15 builds. Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: WARNING:core:fm_malloc: not enough contiguous free pkg memory (317724640 bytes left, need 33832), attempting defragmentation... please increase the "-M" command line parameter! Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: ERROR:core:fm_malloc: not enough free pkg memory (317724640 bytes left, need 33832), please increase the "-M" command line parameter! Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: ERROR:core:receive_msg: no pkg mem left for sip_msg Surprisingly, it happens only with one or two UDP workers of 15 or more. I've already created issue https://github.com/OpenSIPS/opensips/issues/3235 and I'm waiting for it. Meanwhile, I'm looking for a way of restarting particular opensips process. Is it possible? -- Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com From bogdan at opensips.org Thu Nov 9 11:52:42 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 9 Nov 2023 13:52:42 +0200 Subject: [OpenSIPS-Users] OpenSIPS Control Panel 9.3.4 released Message-ID: <2270f24d-174b-e903-b55d-0e5679e151fa@opensips.org> The OpenSIPS Control Panel 9.3.4 is the corresponding version for OpenSIPs 3.4.x . Besides fixes, the main changes in this version are related to the compatibility to OpenSIPS 3.4, in `dispatcher` and `dialog` tools. http://controlpanel.opensips.org/download.php Enjoy it, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com From marcin at voipplus.net Thu Nov 9 12:03:02 2023 From: marcin at voipplus.net (Marcin Groszek) Date: Thu, 9 Nov 2023 06:03:02 -0600 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> Message-ID: <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> You may try adding more pgk memory in /etc/sysconfig/opensips i.e. P_MEMORY=256 On 11/9/2023 5:07 AM, Alexander Kogan wrote: > Hello all, > > I regularly get a memory error after upgrading to 3.2.14 and 3.2.15 > builds. > > Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: > WARNING:core:fm_malloc: not enough contiguous free pkg memory > (317724640 bytes left, need 33832), attempting defragmentation... > please increase the "-M" command line parameter! > Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: ERROR:core:fm_malloc: > not enough free pkg memory (317724640 bytes left, need 33832), please > increase the "-M" command line parameter! > Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: > ERROR:core:receive_msg: no pkg mem left for sip_msg > > Surprisingly, it happens only with one or two UDP workers of 15 or > more. I've already created issue > https://github.com/OpenSIPS/opensips/issues/3235 and I'm waiting for it. > > Meanwhile, I'm looking for a way of restarting particular opensips > process. Is it possible? > -- Best Regards: Marcin Groszek Business Phone Service https://www.voipplus.net From akogan at 5gfuture.com Thu Nov 9 12:28:02 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Thu, 9 Nov 2023 16:28:02 +0400 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> Message-ID: Hi, of course I did. It doesn't help. Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com On 09.11.2023 16:03, Marcin Groszek wrote: > You may try adding more pgk memory in > > /etc/sysconfig/opensips i.e. > > P_MEMORY=256 > > > On 11/9/2023 5:07 AM, Alexander Kogan wrote: >> Hello all, >> >> I regularly get a memory error after upgrading to 3.2.14 and 3.2.15 >> builds. >> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >> WARNING:core:fm_malloc: not enough contiguous free pkg memory >> (317724640 bytes left, need 33832), attempting defragmentation... >> please increase the "-M" command line parameter! >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: ERROR:core:fm_malloc: >> not enough free pkg memory (317724640 bytes left, need 33832), please >> increase the "-M" command line parameter! >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >> ERROR:core:receive_msg: no pkg mem left for sip_msg >> >> Surprisingly, it happens only with one or two UDP workers of 15 or >> more. I've already created issue >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm waiting for it. >> >> Meanwhile, I'm looking for a way of restarting particular opensips >> process. Is it possible? >> From social at bohboh.info Thu Nov 9 14:12:37 2023 From: social at bohboh.info (Social Boh) Date: Thu, 9 Nov 2023 09:12:37 -0500 Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio Message-ID: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> Hello list, I have a problem in communication between an OpenSIPs and a Kamailio. The call comes from OpenSIPs to Kamailio, it is answered but the ACK that OpenSIPs sends to Kamailio I think is not correct: 200OK from Kamailio to OpenSIPs: Contact: ACK from OpenSIPs to Kamailio: Contact: I think the ACK Contact Header from OpenSIPs to Kamailio should have the same 200 OK content 194.195.XXX.XXX is a Asterisk PBX 177.242.XXX.XXX is OpenSIPs Result Kamailio don't send ACK to Asterisk PBX and the call ends about 30 seconds. Any hint, please? -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Thu Nov 9 14:17:17 2023 From: abalashov at evaristesys.com (Alex Balashov) Date: Thu, 9 Nov 2023 09:17:17 -0500 Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio In-Reply-To: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> References: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> Message-ID: <3D8ADD70-2650-4B7F-9AF2-9619970FDAD0@evaristesys.com> Hi, 1) Neither Kamailio nor OpenSIPS send 200 OKs; 2) Neither Kamailio nor OPenSIPS send ACKs. They merely relay these. 3) Contact URI alterations may be occurring along the chain, and are likely causing your issue. -- Alex > On 9 Nov 2023, at 09:12, Social Boh wrote: > > Hello list, > I have a problem in communication between an OpenSIPs and a Kamailio. The call comes from OpenSIPs to Kamailio, it is answered but the ACK that OpenSIPs sends to Kamailio I think is not correct: > 200OK from Kamailio to OpenSIPs: Contact: > ACK from OpenSIPs to Kamailio: Contact: > I think the ACK Contact Header from OpenSIPs to Kamailio should have the same 200 OK content > 194.195.XXX.XXX is a Asterisk PBX > 177.242.XXX.XXX is OpenSIPs > Result Kamailio don't send ACK to Asterisk PBX and the call ends about 30 seconds. > Any hint, please? > -- > --- > I'm SoCIaL, MayBe > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 From solarmon at one-n.co.uk Thu Nov 9 14:24:48 2023 From: solarmon at one-n.co.uk (solarmon) Date: Thu, 9 Nov 2023 14:24:48 +0000 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> Message-ID: What memory management and allocator are you using? I had similar issues using opensips 3.2.7 and the quick workaround was to change from F_MALLOC to HP_MALLOC. But I thought there should have been fixes for such issues in newer versions, so your issue might be different. On Thu, 9 Nov 2023 at 12:33, Alexander Kogan wrote: > Hi, > > of course I did. It doesn't help. > > Best regards, > Alexander Kogan, > Director of R&D > 5g Future > http://5gfuture.com > > > On 09.11.2023 16:03, Marcin Groszek wrote: > > You may try adding more pgk memory in > > > > /etc/sysconfig/opensips i.e. > > > > P_MEMORY=256 > > > > > > On 11/9/2023 5:07 AM, Alexander Kogan wrote: > >> Hello all, > >> > >> I regularly get a memory error after upgrading to 3.2.14 and 3.2.15 > >> builds. > >> > >> Nov 9 11:53:02 FI173 /usr/sbin/opensips[3243]: > >> WARNING:core:fm_malloc: not enough contiguous free pkg memory > >> (317724640 bytes left, need 33832), attempting defragmentation... > >> please increase the "-M" command line parameter! > >> Nov 9 11:53:02 FI173 /usr/sbin/opensips[3243]: ERROR:core:fm_malloc: > >> not enough free pkg memory (317724640 bytes left, need 33832), please > >> increase the "-M" command line parameter! > >> Nov 9 11:53:02 FI173 /usr/sbin/opensips[3243]: > >> ERROR:core:receive_msg: no pkg mem left for sip_msg > >> > >> Surprisingly, it happens only with one or two UDP workers of 15 or > >> more. I've already created issue > >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm waiting for > it. > >> > >> Meanwhile, I'm looking for a way of restarting particular opensips > >> process. Is it possible? > >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Thu Nov 9 14:32:28 2023 From: social at bohboh.info (Social Boh) Date: Thu, 9 Nov 2023 09:32:28 -0500 Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio In-Reply-To: <3D8ADD70-2650-4B7F-9AF2-9619970FDAD0@evaristesys.com> References: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> <3D8ADD70-2650-4B7F-9AF2-9619970FDAD0@evaristesys.com> Message-ID: Thank you Alex I'll search to see where the problem may be --- I'm SoCIaL, MayBe El 9/11/2023 a las 9:17 a. m., Alex Balashov escribió: > Hi, > > 1) Neither Kamailio nor OpenSIPS send 200 OKs; > > 2) Neither Kamailio nor OPenSIPS send ACKs. > > They merely relay these. > > 3) Contact URI alterations may be occurring along the chain, and are likely causing your issue. > > -- Alex > >> On 9 Nov 2023, at 09:12, Social Boh wrote: >> >> Hello list, >> I have a problem in communication between an OpenSIPs and a Kamailio. The call comes from OpenSIPs to Kamailio, it is answered but the ACK that OpenSIPs sends to Kamailio I think is not correct: >> 200OK from Kamailio to OpenSIPs: Contact: >> ACK from OpenSIPs to Kamailio: Contact: >> I think the ACK Contact Header from OpenSIPs to Kamailio should have the same 200 OK content >> 194.195.XXX.XXX is a Asterisk PBX >> 177.242.XXX.XXX is OpenSIPs >> Result Kamailio don't send ACK to Asterisk PBX and the call ends about 30 seconds. >> Any hint, please? >> -- >> --- >> I'm SoCIaL, MayBe >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From akogan at 5gfuture.com Thu Nov 9 14:32:35 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Thu, 9 Nov 2023 18:32:35 +0400 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> Message-ID: I've tried all of them - Q, F, and HP with the same result. Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com On 09.11.2023 18:24, solarmon wrote: > What memory management and allocator are you using? > > I had similar issues using opensips 3.2.7 and the quick workaround was > to change from F_MALLOC to HP_MALLOC. But I thought there should have > been fixes for such issues in newer versions, so your issue might be > different. > > On Thu, 9 Nov 2023 at 12:33, Alexander Kogan wrote: > > Hi, > > of course I did. It doesn't help. > > Best regards, > Alexander Kogan, > Director of R&D > 5g Future > http://5gfuture.com > > > On 09.11.2023 16:03, Marcin Groszek wrote: > > You may try adding more pgk memory in > > > > /etc/sysconfig/opensips i.e. > > > > P_MEMORY=256 > > > > > > On 11/9/2023 5:07 AM, Alexander Kogan wrote: > >> Hello all, > >> > >> I regularly get a memory error after upgrading to 3.2.14 and > 3.2.15 > >> builds. > >> > >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: > >> WARNING:core:fm_malloc: not enough contiguous free pkg memory > >> (317724640 bytes left, need 33832), attempting defragmentation... > >> please increase the "-M" command line parameter! > >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: > ERROR:core:fm_malloc: > >> not enough free pkg memory (317724640 bytes left, need 33832), > please > >> increase the "-M" command line parameter! > >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: > >> ERROR:core:receive_msg: no pkg mem left for sip_msg > >> > >> Surprisingly, it happens only with one or two UDP workers of 15 or > >> more. I've already created issue > >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm > waiting for it. > >> > >> Meanwhile, I'm looking for a way of restarting particular opensips > >> process. Is it possible? > >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Nov 9 14:51:33 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 9 Nov 2023 14:51:33 +0000 Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio In-Reply-To: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> References: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> Message-ID: The Contact fields in messages sent from different User Agents are not expected to be the same. The Contact header provides a URL which can be used to contact the UA about the dialog. If UAs were sending the same Contact as other UAs, they would be directing future requests to a different server. It is the Request URI of the ACK that should match the Contact from the 200 OK. The ACK is not required to have a Contact at all, as the UAC’s Contact was already provided in the INVITE. Ben Newlin From: Users on behalf of Social Boh Date: Thursday, November 9, 2023 at 9:16 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello list, I have a problem in communication between an OpenSIPs and a Kamailio. The call comes from OpenSIPs to Kamailio, it is answered but the ACK that OpenSIPs sends to Kamailio I think is not correct: 200OK from Kamailio to OpenSIPs: Contact: ACK from OpenSIPs to Kamailio: Contact: I think the ACK Contact Header from OpenSIPs to Kamailio should have the same 200 OK content 194.195.XXX.XXX is a Asterisk PBX 177.242.XXX.XXX is OpenSIPs Result Kamailio don't send ACK to Asterisk PBX and the call ends about 30 seconds. Any hint, please? -- --- I'm SoCIaL, MayBe -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Thu Nov 9 16:39:32 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Thu, 9 Nov 2023 08:39:32 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: I created a dummy SDP to pass to RTPEngine using the optional body string "*body(string, optional) - used to provide a specific body to the rtpengine_* function. If this parameter is missing the body of the current message is used.*" and RTPengine was able to create the response SDP. It did, however, show me that the ACK was still not being accepted from the original 200OK as I am now getting the ACK for both CSeq now. So I have two issues, the ACK is not being accepted, and the no SDP on the late media INVITE. Not sure what is the best way to handle the Late Media INVITE with RTPEngine as so far it has only worked when it receives an SDP. This is the snippit of code that I am using for this and using the IP of the Opensips server and an arbitrary port in the SDP creation. $var(newbody) = "v=0\r\no=Opensips 1 IN IP4 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 3140 RTP/AVP 0 101\r\na=sendrecv\r\ na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\n"; xlog("New body is $var(newbody)"); create_dialog(); rtpengine_offer("from-tag=$ft replace-session-connection trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); xlog("Body from RTPENGINE is $var(body)"); $json(reply) := $rtpquery; $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); append_to_reply("Contact:\r\n"); append_to_reply("Content-Type: application/sdp\r\n"); xlog("Body is $var(body)"); t_reply_with_body(200, "OK", $var(body)); rtpengine_play_media("call-id=$ci from-tag=$ft file=/etc/rtpengine/unk_num.wav"); async(sleep(10), after_media); Here is the sngrep showing that first ACK is coming in with CSEQ 1 for example, Reinvite, 200OK and ack for CSEQ 2, then 200OK and ACK for CSEQ 1 then 200OK and ACK for CSEQ 2. until it times out with the async(sleep(10), after_media); Any help is appreciated Thank you Kevin On Tue, Nov 7, 2023 at 9:19 AM Kevin Kennedy wrote: > Looks like in the debug it is not finding an SDP so it cant construct a > response. Is there a way to remedy this? > > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:rtpengine:extract_body: No body found > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > ERROR:rtpengine:rtpengine_offer_answer_body: can't extract body from the > message > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=40 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_to_param: tag=1985761840-1699375436084- > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_to_param: end of header reached, state=11 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:_parse_to: end of header reached, state=29 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:_parse_to: display={}, ruri={sip:+17024054893 at 10.20.252.101 > ;user=phone} > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:rtpengine:rtpe_function_call: proxy reply: > d7:createdi1699375436e10:created_usi125586e11:last > signali1699375436e4:SSRCde4:tagsde6:totalsd3:RTPd7:packetsi0e5:bytesi0e6:errorsi0ee4:RTCPd7:packetsi0e5:bytesi0e6:errorsi0eee6:result2:oke > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=ffffffffffffffff > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:sipmsgops:remove_body_part_f: no body found > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=ffffffffffffffff > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:pv_get_msg_body: no message body > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > ERROR:core:get_cmd_fixups: Variable in param [3] is not a string > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > ERROR:core:do_action: Failed to get fixups for command > in /etc/opensips/opensips.cfg, line 284 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:rtpengine:rtpe_function_call: proxy reply: > d8:durationi10152e6:result2:oke > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:t_newtran: transaction on entrance=(nil) > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=ffffffffffffffff > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=78 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:t_lookup_request: start searching: hash=31608, isACK=0 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:matching_3261: RFC3261 transaction matching failed > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:t_lookup_request: no transaction found > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:run_any_trans_callbacks: trans=0x7f920581abe8, callback type 1, id 0 > entered > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:core:parse_headers: flags=ffffffffffffffff > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:_reply_light: reply sent out. buf=0x7f92094d8218: SIP/2.0 1..., > shmem=0x7f920581dfa8: SIP/2.0 1 > Nov 7 11:43:56 lab-opensips opensips[4670]: Nov 7 11:43:56 [4670] > DBG:tm:_reply_light: finished > > Thank you. > > Kevin > > On Mon, Nov 6, 2023 at 2:58 PM Kevin Kennedy > wrote: > >> I would like to clarify the issue in case its not 100% clear. >> * Caller sends INVITE with No SDP(Late Media Invite) >> * Another device in path (B2BUA) receives INVITE and sends dummy SDP to >> Opensips with just G.711 codec in the Offer >> * Opensips Creates Dialog and sends 200OK with SDP using >> t_reply_with_body based on previously provided information. >> * B2BUA receives 200OK with SDP then sends ACK followed by a Re-INVITE >> with No SDP back to OpenSips. >> * Opensips appears to accept the ACK as it doesn't retransmit the 200OK >> right away as before updated changes. >> * Opensips sends 100 trying with new CSEQ from Re-INVITE with no SDP >> * 200OK Loop created >> * Opensips send 200 OK with old CSEQ >> * B2BUA sends ACK with old CSEQ >> * Call times out. >> >> No audio sent >> >> Thank you >> >> Kevin. >> * >> >> >> >> On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy >> wrote: >> >>> I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there >>> was some re-invite fixes. Still doesn't seem to resolve this issue. What >>> am I missing to handle this correctly? >>> >>> Thank you. >>> >>> Kevin >>> >>> On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy >>> wrote: >>> >>>> Dmitry, >>>> Thank you for your response, it does appear to work this way and is >>>> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly >>>> with the updated Cseq in the 100 Trying, but the 200 OK (using the >>>> t_reply_with_body), still has the same Cseq as the initial INVITE. How can >>>> I make adjustments for this? >>>> >>>> Thank you. >>>> >>>> Kevin >>>> >>>> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov >>>> wrote: >>>> >>>>> It turns out that this is no early_media, there were simply successful >>>>> attempts with 183 Session Progress, which is why there was such a >>>>> misunderstanding, I’ll attach the snippet code again in plain text: >>>>> route { if (is_method("INVITE")) { create_dialog(); route(media); >>>>> exit; >>>>> } } route[media] { if (has_body("application/sdp")) { >>>>> rtpengine_offer(); >>>>> } $json(reply) := $rtpquery; >>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local >>>>> port); >>>>> remove_body_part(); append_to_reply("Contact: >>>>> \r\n"); >>>>> >>>>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>>>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>> t_reply_with_body(200, "OK", $var(body)); >>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } >>>>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); >>>>> exit; >>>>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } } >>>>> >>>>> and pined previous posts below :) >>>>> >>>>> > >>>>> ---------------------------------------------------------------------- >>>>> > Message: 2 >>>>> > Date: Fri, 3 Nov 2023 16:00:22 +0500 >>>>> > From: Dmitry Ponomaryov >>>>> > To:users at lists.opensips.org >>>>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>>>> > absorbing ACK >>>>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8b5a at gmail.com> >>>>> > Content-Type: text/plain; charset="utf-8"; Format="flowed" >>>>> > >>>>> > Hello everyone, I would like to show my part of the code when playing >>>>> > early media after 200OK, when creating dialogs, I substituted >>>>> $DLG_did >>>>> > in the contact of my dialog, and received the same $DLG_did for my >>>>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite >>>>> > having already received an ACK response. >>>>> > >>>>> > route { >>>>> > >>>>> > # initial invite >>>>> > >>>>> > if (is_method("INVITE")) { >>>>> > >>>>> > create_dialog(); >>>>> > >>>>> > route(early_media); >>>>> > >>>>> > exit; >>>>> > >>>>> > } >>>>> > >>>>> > } route[early_media] { if (has_body("application/sdp")) { >>>>> > rtpengine_manage(); } $json(reply) := $rtpquery; >>>>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local >>>>> port); >>>>> > remove_body_part(); >>>>> > >>>>> > append_to_reply("Contact: >>>>> > >>>> $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n"); >>>>> > >>>>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = >>>>> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = >>>>> > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>> > t_reply_with_body(200, "OK", $var(body)); >>>>> > rtpengine_play_media("call-id=$ci from-tag=$ft >>>>> > file=/etc/rtpengine/media.wav"); async(sleep(10), >>>>> after_early_media); } >>>>> > route[after_early_media] { if (t_was_cancelled()) { >>>>> rtpengine_delete(); >>>>> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); >>>>> exit; >>>>> > } } >>>>> > >>>>> > I don’t know if Kevin example was with creating a dialog, but I also >>>>> > noticed this problem through transaction... thanks >>>>> > -------------- next part -------------- >>>>> > An HTML attachment was scrubbed... >>>>> > URL:< >>>>> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html >>>>> > >>>>> > >>>>> ---------------------------------------------------------------------- >>>>> > >>>>> > Message: 1 >>>>> > Date: Thu, 2 Nov 2023 16:32:02 -0700 >>>>> > From: Kevin Kennedy >>>>> > To: OpenSIPS users mailling list >>>>> > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not >>>>> > absorbing ACK >>>>> > Message-ID: >>>>> > < >>>>> CABDXsRxLTp2_uEX_UPX1adg16af6gaetzJujUTPki8c7H3KKLQ at mail.gmail.com> >>>>> > Content-Type: text/plain; charset="utf-8" >>>>> > >>>>> > I am trying to build a solution where Opensips 3.2+ with RTPengine >>>>> acts as >>>>> > a UAC, answers a call with 200OK, plays media from file, and will >>>>> terminate >>>>> > the call right after playing announcement. >>>>> > >>>>> > Opensips is responding with 200OK with SDP body and making the >>>>> > correct changes for the IP, but when the ACK comes back from the UAS, >>>>> > Opensips doesn't seem to absorb it and retransmits the 200OK. >>>>> > >>>>> > Code snippet handling this scenario >>>>> > >>>>> > rtpengine_manage("from-tag=$ft replace-session-connection >>>>> > trust-address replace-origin codec-strip-g729",,$var(body)); >>>>> > append_to_reply("Contact:\r\n"); >>>>> > append_to_reply("Content-Type: application/sdp\r\n"); >>>>> > t_reply_with_body(200, "OK", $var(body)); >>>>> > rtpengine_play_media("from-tag=$ft >>>>> > file=/etc/rtpengine/unk_num.wav"); >>>>> > sleep(10); >>>>> > rtpengine_delete("from-tag=$ft"); >>>>> > #t_reply(603, "Decline"); >>>>> > exit(); >>>>> > >>>>> > >>>>> > What do I need to add to handle this scenario correctly? >>>>> > >>>>> > Note: I was able to get this to work with Early Media (183 >>>>> > reply_with_body, and send t_reply(603, "Decline")), but we have >>>>> customers >>>>> > using late media invite as well, so the Early Media option wouldn't >>>>> work in >>>>> > that case. >>>>> > >>>>> > Thank you. >>>>> > >>>>> > Kevin Kennedy >>>>> > -------------- next part -------------- >>>>> > An HTML attachment was scrubbed... >>>>> > URL:< >>>>> http://lists.opensips.org/pipermail/users/attachments/20231102/dd52d307/attachment-0001.html >>>>> > >>>>> > >>>>> > ------------------------------ >>>>> > >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Fri Nov 10 19:44:41 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Fri, 10 Nov 2023 11:44:41 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: >>>>>>> I was able to get audio, The problem I was having is the Originator >>>>>>> string in the SDP. However, I am still having the same issue with >>>>>>> accepting the ACK from the Originator and not resending the 200OK. Can >>>>>>> someone please help with this issue? >>>>>>> >>>>>>> Thank you >>>>>>> >>>>>>> *Code snippet for the Late Media route* >>>>>>> route["LateMedia3"]{ >>>>>>> if (has_body("application/sdp")) { >>>>>>> xlog("######## Entered route LateMedia3 with Fake SDP from >>>>>>> Originator ########\r\n"); >>>>>>> rtpengine_offer(); >>>>>>> $json(reply) := $rtpquery; >>>>>>> >>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>> >>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>> remove_body_part(); >>>>>>> append_to_reply("Contact:>>>>>> $socket_in(ip):$socket_in(port);user=phone>\r\n"); >>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>> $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>> $var(body) = >>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>> async(sleep(10), after_media); >>>>>>> } else { >>>>>>> xlog("######## Entered route LateMedia3 No SDP received, >>>>>>> Create one from variable ########\r\n"); >>>>>>> $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 >>>>>>> 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 3140 >>>>>>> RTP/AVP 0 101\r\na >>>>>>> =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >>>>>>> telephone-event/8000\r\na=fmtp:101 0-15\r\n"); >>>>>>> xlog("######################### Body to RTPENGINE is >>>>>>> ###########################\r\n$var(newbody)\r\n"); >>>>>>> rtpengine_offer("from-tag=$ft replace-session-connection >>>>>>> trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); >>>>>>> xlog("######################### Body from RTPENGINE is >>>>>>> ###########################\r\n$var(body)\r\n"); >>>>>>> $json(reply) := $rtpquery; >>>>>>> >>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>> >>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>> append_to_reply("Contact:>>>>>> $socket_in(ip):$socket_in(port);transport=udp>\r\n"); >>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>> $var(body) = $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>> $var(body) = >>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>> xlog("######################### Body being sent in Reply is >>>>>>> ######################\r\n$var(body)\r\n"); >>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>> async(sleep(10), after_media); >>>>>>> } >>>>>>> } >>>>>>> >>>>>>> route[after_media] >>>>>>> { if (t_was_cancelled()) { >>>>>>> rtpengine_delete(); >>>>>>> exit; >>>>>>> } else { >>>>>>> rtpengine_delete(); >>>>>>> sl_send_reply(486,"Busy here"); >>>>>>> exit; >>>>>>> } >>>>>>> } >>>>>>> >>>>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Sat Nov 11 00:54:56 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Fri, 10 Nov 2023 16:54:56 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: Looks like if I put t_newtran(); in the main route this created the transaction and allowed the ACK to be recognized. Now How do I force Opensips to send a BYE. Thank you. On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy wrote: > > > >>>>>>>> I was able to get audio, The problem I was having is the >>>>>>>> Originator string in the SDP. However, I am still having the same issue >>>>>>>> with accepting the ACK from the Originator and not resending the 200OK. >>>>>>>> Can someone please help with this issue? >>>>>>>> >>>>>>>> Thank you >>>>>>>> >>>>>>>> *Code snippet for the Late Media route* >>>>>>>> route["LateMedia3"]{ >>>>>>>> if (has_body("application/sdp")) { >>>>>>>> xlog("######## Entered route LateMedia3 with Fake SDP from >>>>>>>> Originator ########\r\n"); >>>>>>>> rtpengine_offer(); >>>>>>>> $json(reply) := $rtpquery; >>>>>>>> >>>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>>> >>>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>>> remove_body_part(); >>>>>>>> append_to_reply("Contact:>>>>>>> $socket_in(ip):$socket_in(port);user=phone>\r\n"); >>>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>>> $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>>> $var(body) = >>>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>>> async(sleep(10), after_media); >>>>>>>> } else { >>>>>>>> xlog("######## Entered route LateMedia3 No SDP received, >>>>>>>> Create one from variable ########\r\n"); >>>>>>>> $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 >>>>>>>> 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 3140 >>>>>>>> RTP/AVP 0 101\r\na >>>>>>>> =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >>>>>>>> telephone-event/8000\r\na=fmtp:101 0-15\r\n"); >>>>>>>> xlog("######################### Body to RTPENGINE is >>>>>>>> ###########################\r\n$var(newbody)\r\n"); >>>>>>>> rtpengine_offer("from-tag=$ft replace-session-connection >>>>>>>> trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); >>>>>>>> xlog("######################### Body from RTPENGINE is >>>>>>>> ###########################\r\n$var(body)\r\n"); >>>>>>>> $json(reply) := $rtpquery; >>>>>>>> >>>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>>> >>>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>>> append_to_reply("Contact:>>>>>>> $socket_in(ip):$socket_in(port);transport=udp>\r\n"); >>>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>>> $var(body) = >>>>>>>> $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>>> $var(body) = >>>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>>> xlog("######################### Body being sent in Reply is >>>>>>>> ######################\r\n$var(body)\r\n"); >>>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>>> async(sleep(10), after_media); >>>>>>>> } >>>>>>>> } >>>>>>>> >>>>>>>> route[after_media] >>>>>>>> { if (t_was_cancelled()) { >>>>>>>> rtpengine_delete(); >>>>>>>> exit; >>>>>>>> } else { >>>>>>>> rtpengine_delete(); >>>>>>>> sl_send_reply(486,"Busy here"); >>>>>>>> exit; >>>>>>>> } >>>>>>>> } >>>>>>>> >>>>>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From kennedy4260 at gmail.com Sat Nov 11 01:26:30 2023 From: kennedy4260 at gmail.com (Kevin Kennedy) Date: Fri, 10 Nov 2023 17:26:30 -0800 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: I was able to send the BYE to the call by adding a parameter in the dialog module to timeout the dialog with a short time letting the announcement play, and added the create_dialog with the flag of B to send BYE on dialog timeout at the beginning of the route. Now that the transactions are working correctly, I can use the same route for the calls with SDP as well and tighten up the script. Thanks for helping out with some code examples, and letting me update on my progress on this thread. Hopefully this can help someone else out having a similar problem when trying to use Opensips with RTPENGINE as an announcement server. modparam("dialog", "default_timeout", 12) route["RTPENGINE"]{ if (has_body("application/sdp")) { create_dialog("B"); rtpengine_offer(); $json(reply) := $rtpquery; $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); remove_body_part(); append_to_reply("Contact:\r\n"); append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); $var(body) = $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); t_reply_with_body(200, "OK", $var(body)); rtpengine_play_media("call-id=$ci from-tag=$ft file=/etc/rtpengine/unk_num.wav"); exit; } else { create_dialog("B"); $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 " + $socket_in(ip) + "\r\ns=-\r\nc=IN IP4 " + $socket_in(ip) + "\r\nt=0 0\r\nm=audio " + $sp + " RTP/AVP 0 101\r\na=sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\n"); rtpengine_offer("from-tag=$ft replace-session-connection trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); $json(reply) := $rtpquery; $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); append_to_reply("Contact: wrote: > Looks like if I put t_newtran(); in the main route this created the > transaction and allowed the ACK to be recognized. Now How do I force > Opensips to send a BYE. > > Thank you. > > On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy > wrote: > >> >> >> >>>>>>>>> I was able to get audio, The problem I was having is the >>>>>>>>> Originator string in the SDP. However, I am still having the same issue >>>>>>>>> with accepting the ACK from the Originator and not resending the 200OK. >>>>>>>>> Can someone please help with this issue? >>>>>>>>> >>>>>>>>> Thank you >>>>>>>>> >>>>>>>>> *Code snippet for the Late Media route* >>>>>>>>> route["LateMedia3"]{ >>>>>>>>> if (has_body("application/sdp")) { >>>>>>>>> xlog("######## Entered route LateMedia3 with Fake SDP from >>>>>>>>> Originator ########\r\n"); >>>>>>>>> rtpengine_offer(); >>>>>>>>> $json(reply) := $rtpquery; >>>>>>>>> >>>>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>>>> >>>>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>>>> remove_body_part(); >>>>>>>>> append_to_reply("Contact:>>>>>>>> $socket_in(ip):$socket_in(port);user=phone>\r\n"); >>>>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>>>> $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>>>> $var(body) = >>>>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>>>> async(sleep(10), after_media); >>>>>>>>> } else { >>>>>>>>> xlog("######## Entered route LateMedia3 No SDP received, >>>>>>>>> Create one from variable ########\r\n"); >>>>>>>>> $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 >>>>>>>>> 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 3140 >>>>>>>>> RTP/AVP 0 101\r\na >>>>>>>>> =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101 >>>>>>>>> telephone-event/8000\r\na=fmtp:101 0-15\r\n"); >>>>>>>>> xlog("######################### Body to RTPENGINE is >>>>>>>>> ###########################\r\n$var(newbody)\r\n"); >>>>>>>>> rtpengine_offer("from-tag=$ft replace-session-connection >>>>>>>>> trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); >>>>>>>>> xlog("######################### Body from RTPENGINE is >>>>>>>>> ###########################\r\n$var(body)\r\n"); >>>>>>>>> $json(reply) := $rtpquery; >>>>>>>>> >>>>>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); >>>>>>>>> >>>>>>>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local address); >>>>>>>>> append_to_reply("Contact:>>>>>>>> $socket_in(ip):$socket_in(port);transport=udp>\r\n"); >>>>>>>>> append_to_reply("Content-Type: application/sdp\r\n"); >>>>>>>>> $var(body) = >>>>>>>>> $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g}); >>>>>>>>> $var(body) = >>>>>>>>> $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >>>>>>>>> xlog("######################### Body being sent in Reply >>>>>>>>> is ######################\r\n$var(body)\r\n"); >>>>>>>>> t_reply_with_body(200, "OK", $var(body)); >>>>>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft >>>>>>>>> file=/etc/rtpengine/unk_num.wav"); >>>>>>>>> async(sleep(10), after_media); >>>>>>>>> } >>>>>>>>> } >>>>>>>>> >>>>>>>>> route[after_media] >>>>>>>>> { if (t_was_cancelled()) { >>>>>>>>> rtpengine_delete(); >>>>>>>>> exit; >>>>>>>>> } else { >>>>>>>>> rtpengine_delete(); >>>>>>>>> sl_send_reply(486,"Busy here"); >>>>>>>>> exit; >>>>>>>>> } >>>>>>>>> } >>>>>>>>> >>>>>>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 07:40:37 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 09:40:37 +0200 Subject: [OpenSIPS-Users] Need some help in drop() core function of opensips . In-Reply-To: References: Message-ID: <8cc7bc88-9160-3176-221a-7e03cffc62e6@opensips.org> Hi, 1) In branch route you cannot do signaling (like sending a reply). Is the 500 reply generated from somewhere from your script? as OpenSIPS is may not send something like that by default. Maybe dropping the branch makes the t_relay() to fail and you have the 500 sending there ? 2) just print in branch route as $ru. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/8/23 3:08 PM, Sasmita Panda wrote: > Hi All , > > branch_route[1] { >   xlog("L_NOTICE", " Branch route URI :  $(branch(uri)) branch index : > $T_branch_idx \n"); > >  xlog(" current branch Q value : $(branch(q))  "); >            if (isbflagset("Invalid")){ >             xlog(" dropping current branch  : > $(branch(uri)[$T_branch_idx])  "); > >                  drop; > } > } > > 1. When drop() is called , it sends *"500 Server error occurred" . I > want a custom  error response on this . Is this possible ?  If yes , > then how will I do this ? * > * > * > *2.* Inside the branch route I wanted to print the*branch uri and q > value* , but it's giving me NULL . Am I doing something wrong here ? > * > * > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 07:43:06 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 09:43:06 +0200 Subject: [OpenSIPS-Users] memcached opensips 3.3 In-Reply-To: References: Message-ID: Hi Jason, you say the failover (inside the group) works ok when using REDIS, but not with MemCached ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/3/23 2:04 AM, nz deals wrote: > After additional testing, I observed that the settings for Redis are > functioning properly, just for your information. It appears there > might be an issue specifically when using memcached in a group. > > Thanks > > Regards, > Jason > > On Thu, 2 Nov 2023 at 21:09, nz deals > wrote: > > Hope everyone is having a good day, > I've encountered an issue related to making a memcached group. My > understanding is that when configuring a group, OpenSIPS attempts > to connect to other memcached servers if one is unresponsive. > However, it seems to only connect to the first server in the group > and perform insertions exclusively on that one. In the event that > the first server becomes inaccessible, I receive an error message > stating "Failed to insert: CONNECTION FAILURE" and no > connection/insertion to the second one. > > Here is a snippet of the configuration I'm using: > modparam("cachedb_memcached", > "cachedb_url","memcached:main://memcacheserver1:11222,memcachedserver2:11222/") > > I've thoroughly reviewed the documentation available, but I > couldn't find sufficient clarity on this behavior. I would greatly > appreciate it if someone could provide insights or clarification > on this matter. > > Thanks > Jason > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 07:50:04 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 09:50:04 +0200 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> Message-ID: <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> Hi Alexander, The warning is actually reporting a highly fragmented memory and not a lack of memory. It reports 303 available Mb, but not a single continuous slot for like 33K. By doing an MI "ps", could you identify the name/type of the process reporting this issue ? do you get it from a single process or from multiple processes ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/9/23 4:32 PM, Alexander Kogan wrote: > > I've tried all of them - Q, F, and HP with the same result. > > Best regards, > Alexander Kogan, > Director of R&D317724640 > 5g Future > http://5gfuture.com > > > On 09.11.2023 18:24, solarmon wrote: >> What memory management and allocator are you using? >> >> I had similar issues using opensips 3.2.7 and the quick workaround >> was to change from F_MALLOC to HP_MALLOC. But I thought there should >> have been fixes for such issues in newer versions, so your issue >> might be different. >> >> On Thu, 9 Nov 2023 at 12:33, Alexander Kogan wrote: >> >> Hi, >> >> of course I did. It doesn't help. >> >> Best regards, >> Alexander Kogan, >> Director of R&D >> 5g Future >> http://5gfuture.com >> >> >> On 09.11.2023 16:03, Marcin Groszek wrote: >> > You may try adding more pgk memory in >> > >> > /etc/sysconfig/opensips i.e. >> > >> > P_MEMORY=256 >> > >> > >> > On 11/9/2023 5:07 AM, Alexander Kogan wrote: >> >> Hello all, >> >> >> >> I regularly get a memory error after upgrading to 3.2.14 and >> 3.2.15 >> >> builds. >> >> >> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >> >> WARNING:core:fm_malloc: not enough contiguous free pkg memory >> >> (317724640 bytes left, need 33832), attempting defragmentation... >> >> please increase the "-M" command line parameter! >> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >> ERROR:core:fm_malloc: >> >> not enough free pkg memory (317724640 bytes left, need 33832), >> please >> >> increase the "-M" command line parameter! >> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >> >> ERROR:core:receive_msg: no pkg mem left for sip_msg >> >> >> >> Surprisingly, it happens only with one or two UDP workers of >> 15 or >> >> more. I've already created issue >> >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm >> waiting for it. >> >> >> >> Meanwhile, I'm looking for a way of restarting particular >> opensips >> >> process. Is it possible? >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 07:57:14 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 09:57:14 +0200 Subject: [OpenSIPS-Users] Need some help in lookup flag "B" in opensips 3.2 version . In-Reply-To: References: Message-ID: <948cf01a-76f3-a6a9-1f58-74ccc17550a7@opensips.org> Hi, May I ask why using the "B" flag here ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/7/23 9:00 AM, Sasmita Panda wrote: > Hi All , > > I am using opensips 3.2 version . > route { > ------- >      if ($rm=="INVITE") >                 { >                    if(!lookup("location","B")) >                       { >                        if (!t_reply(404, "Not Found")) >                         { >                             sl_reply_error(); >                         } >                         exit; >                       } >                 } >                         if (!serialize_branches(1)){ >                                 sl_send_reply(500,"Unable to load > contacts"); >                                 exit; >                         }else{ >                            #     if (next_branches()){ >                                         t_on_failure("1"); >                                } >         } > > Then of course the route(1) and failure_route(1) i have called . But > what is happening in my case is . I have 2 branches , For Invite > opensips tries both the branches but if both branch wont accept the > call Opensips should reply error code to the caller . But Opensips > create a new invite as the Caller sent and again follows the lookup > logic . > > Why is this happening ? How will I manage this ? > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 07:59:02 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 09:59:02 +0200 Subject: [OpenSIPS-Users] Strange ACK between OpenSIPS and Kamailio In-Reply-To: References: <89724693-30ee-435c-a4b9-b81f0db3c632@bohboh.info> <3D8ADD70-2650-4B7F-9AF2-9619970FDAD0@evaristesys.com> Message-ID: Hi, if you have a pcap showing the inbound and outbound traffic on OpenSIPS, I can do a quick doublecheck - of course, please send it off list. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/9/23 4:32 PM, Social Boh wrote: > Thank you Alex > > I'll search to see where the problem may be > > --- > I'm SoCIaL, MayBe > > El 9/11/2023 a las 9:17 a. m., Alex Balashov escribió: >> Hi, >> >> 1) Neither Kamailio nor OpenSIPS send 200 OKs; >> >> 2) Neither Kamailio nor OPenSIPS send ACKs. >> >> They merely relay these. >> >> 3) Contact URI alterations may be occurring along the chain, and are >> likely causing your issue. >> >> -- Alex >> >>> On 9 Nov 2023, at 09:12, Social Boh wrote: >>> >>> Hello list, >>> I have a problem in communication between an OpenSIPs and a >>> Kamailio. The call comes from OpenSIPs to Kamailio, it is answered >>> but the ACK that OpenSIPs sends to Kamailio I think is not correct: >>> 200OK from Kamailio to OpenSIPs: Contact: >>> ACK from OpenSIPs to Kamailio: Contact: >>> >>> I think the ACK Contact Header from OpenSIPs to Kamailio should have >>> the same 200 OK content >>> 194.195.XXX.XXX is a Asterisk PBX >>> 177.242.XXX.XXX is OpenSIPs >>> Result Kamailio don't send ACK to Asterisk PBX and the call ends >>> about 30 seconds. >>> Any hint, please? >>> -- >>> --- >>> I'm SoCIaL, MayBe >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From david.villasmil.work at gmail.com Mon Nov 13 08:23:43 2023 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 13 Nov 2023 09:23:43 +0100 Subject: [OpenSIPS-Users] Need some help in lookup flag "B" in opensips 3.2 version . In-Reply-To: <948cf01a-76f3-a6a9-1f58-74ccc17550a7@opensips.org> References: <948cf01a-76f3-a6a9-1f58-74ccc17550a7@opensips.org> Message-ID: And what do you mean by “opensips creates a new invite as the caller sent”? On Mon, 13 Nov 2023 at 08:57, Bogdan-Andrei Iancu wrote: > Hi, > > May I ask why using the "B" flag here ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/7/23 9:00 AM, Sasmita Panda wrote: > > Hi All , > > I am using opensips 3.2 version . > route { > ------- > if ($rm=="INVITE") > { > if(!lookup("location","B")) > { > if (!t_reply(404, "Not Found")) > { > sl_reply_error(); > } > exit; > } > } > if (!serialize_branches(1)){ > sl_send_reply(500,"Unable to load > contacts"); > exit; > }else{ > # if (next_branches()){ > t_on_failure("1"); > } > } > > Then of course the route(1) and failure_route(1) i have called . But what > is happening in my case is . I have 2 branches , For Invite opensips tries > both the branches but if both branch wont accept the call Opensips should > reply error code to the caller . But Opensips create a new invite as the > Caller sent and again follows the lookup logic . > > Why is this happening ? How will I manage this ? > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Mon Nov 13 09:30:40 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 13 Nov 2023 15:00:40 +0530 Subject: [OpenSIPS-Users] Need some help in lookup flag "B" in opensips 3.2 version . In-Reply-To: References: <948cf01a-76f3-a6a9-1f58-74ccc17550a7@opensips.org> Message-ID: Thank you All . My problem was resolved . I have managed the config to make it work . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Nov 13, 2023 at 1:53 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > And what do you mean by “opensips creates a new invite as the caller sent”? > > On Mon, 13 Nov 2023 at 08:57, Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> May I ask why using the "B" flag here ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/7/23 9:00 AM, Sasmita Panda wrote: >> >> Hi All , >> >> I am using opensips 3.2 version . >> route { >> ------- >> if ($rm=="INVITE") >> { >> if(!lookup("location","B")) >> { >> if (!t_reply(404, "Not Found")) >> { >> sl_reply_error(); >> } >> exit; >> } >> } >> if (!serialize_branches(1)){ >> sl_send_reply(500,"Unable to load >> contacts"); >> exit; >> }else{ >> # if (next_branches()){ >> t_on_failure("1"); >> } >> } >> >> Then of course the route(1) and failure_route(1) i have called . But what >> is happening in my case is . I have 2 branches , For Invite opensips tries >> both the branches but if both branch wont accept the call Opensips should >> reply error code to the caller . But Opensips create a new invite as the >> Caller sent and again follows the lookup logic . >> >> Why is this happening ? How will I manage this ? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 09:36:49 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 11:36:49 +0200 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> Message-ID: <72f1b287-e9d5-0299-5dba-76a6ac139e8e@opensips.org> Quick update here after a discussion with Liviu. It looks (after checking the #3235 issue, that it is the httpd process, responsible for running the MI cmds when JSONRPC is used. And it is about listing dialog, so a command that may produce a huge output. The exact issue you are reporting here was solved some time ago, please check : f91a4aa0c8 - 3.2.12 d4edccc628 - 3.2.13 You should have them if running 3.2.15 - are you sure you are using the right code and binaries (as version) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/13/23 9:50 AM, Bogdan-Andrei Iancu wrote: > Hi Alexander, > > The warning is actually reporting a highly fragmented memory and not a > lack of memory. It reports 303 available Mb, but not a single > continuous slot for like 33K. > > By doing an MI "ps", could you identify the name/type of the process > reporting this issue ? do you get it from a single process or from > multiple processes ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 11/9/23 4:32 PM, Alexander Kogan wrote: >> >> I've tried all of them - Q, F, and HP with the same result. >> >> Best regards, >> Alexander Kogan, >> Director of R&D317724640 >> 5g Future >> http://5gfuture.com >> >> >> On 09.11.2023 18:24, solarmon wrote: >>> What memory management and allocator are you using? >>> >>> I had similar issues using opensips 3.2.7 and the quick workaround >>> was to change from F_MALLOC to HP_MALLOC. But I thought there should >>> have been fixes for such issues in newer versions, so your issue >>> might be different. >>> >>> On Thu, 9 Nov 2023 at 12:33, Alexander Kogan >>> wrote: >>> >>> Hi, >>> >>> of course I did. It doesn't help. >>> >>> Best regards, >>> Alexander Kogan, >>> Director of R&D >>> 5g Future >>> http://5gfuture.com >>> >>> >>> On 09.11.2023 16:03, Marcin Groszek wrote: >>> > You may try adding more pgk memory in >>> > >>> > /etc/sysconfig/opensips i.e. >>> > >>> > P_MEMORY=256 >>> > >>> > >>> > On 11/9/2023 5:07 AM, Alexander Kogan wrote: >>> >> Hello all, >>> >> >>> >> I regularly get a memory error after upgrading to 3.2.14 and >>> 3.2.15 >>> >> builds. >>> >> >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> >> WARNING:core:fm_malloc: not enough contiguous free pkg memory >>> >> (317724640 bytes left, need 33832), attempting >>> defragmentation... >>> >> please increase the "-M" command line parameter! >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> ERROR:core:fm_malloc: >>> >> not enough free pkg memory (317724640 bytes left, need >>> 33832), please >>> >> increase the "-M" command line parameter! >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> >> ERROR:core:receive_msg: no pkg mem left for sip_msg >>> >> >>> >> Surprisingly, it happens only with one or two UDP workers of >>> 15 or >>> >> more. I've already created issue >>> >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm >>> waiting for it. >>> >> >>> >> Meanwhile, I'm looking for a way of restarting particular >>> opensips >>> >> process. Is it possible? >>> >> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Nov 13 09:40:53 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Nov 2023 11:40:53 +0200 Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK In-Reply-To: References: <3971cbc2-7281-2299-4212-7f241e8b8b5a@gmail.com> Message-ID: <98ed90fe-1d6b-63f0-1a49-6f66bb3a20ec@opensips.org> Hi there, trying to maintain a dialog stateful UAS from script level may be something difficult and painful to do. Maybe you should take a look at the UAC/UAS support provided by the b2b_entities module in OpenSIPS 3.4: https://blog.opensips.org/2023/03/22/api-driven-sip-user-agent-end-point-with-opensips-3-4/ Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/11/23 3:26 AM, Kevin Kennedy wrote: > I was able to send the BYE to the call by adding a parameter in the > dialog module to timeout the dialog with a short time letting the > announcement play, and added the create_dialog with the flag of B to > send BYE on dialog timeout at the beginning of the route.  Now that > the transactions are working correctly, I can use the same route for > the calls with SDP as well and tighten up the script.  Thanks for > helping out with some code examples, and letting me update on my > progress on this thread.  Hopefully this can help someone else out > having a similar problem when trying to use Opensips with RTPENGINE as > an announcement server. > > modparam("dialog", "default_timeout", 12) > > route["RTPENGINE"]{ >     if (has_body("application/sdp")) { >         create_dialog("B"); >         rtpengine_offer(); >         $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); > $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > address); >         remove_body_part(); > append_to_reply("Contact:\r\n"); >         append_to_reply("Content-Type: application/sdp\r\n"); >         $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); >         $var(body) = $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >         t_reply_with_body(200, "OK", $var(body)); >         rtpengine_play_media("call-id=$ci from-tag=$ft > file=/etc/rtpengine/unk_num.wav"); >         exit; >     } else { >         create_dialog("B"); >         $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 " + > $socket_in(ip) + "\r\ns=-\r\nc=IN IP4 " + $socket_in(ip) + "\r\nt=0 > 0\r\nm=audio " + $sp + " RTP/AVP 0 101\r\na=sendrecv\r\na=rtpmap:0 > PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\n"); >         rtpengine_offer("from-tag=$ft replace-session-connection > trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody)); >         $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); > $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > address); > append_to_reply("Contact:         append_to_reply("Content-Type: application/sdp\r\n"); >         $var(body) = $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); >         t_reply_with_body(200, "OK", $var(body)); >         rtpengine_play_media("call-id=$ci from-tag=$ft > file=/etc/rtpengine/unk_num.wav"); >         exit; >     } > > } > > Thank you. > > Kevin > > On Fri, Nov 10, 2023 at 4:54 PM Kevin Kennedy > wrote: > > Looks like if I put t_newtran(); in the main route this created > the transaction and allowed the ACK to be recognized.  Now How do > I force Opensips to send a BYE. > > Thank you. > > On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy > > wrote: > > > > > I was able to get audio, The > problem I was having is the > Originator string in the SDP.  > However, I am still having the > same issue with accepting the ACK > from the Originator and not > resending the 200OK.  Can someone > please help with this issue? > > Thank you > > *Code snippet for the Late Media > route* > route["LateMedia3"]{ >     if (has_body("application/sdp")) { >         xlog("######## Entered > route LateMedia3 with Fake SDP > from Originator ########\r\n"); > rtpengine_offer(); >         $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > port); > $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > address); > remove_body_part(); > append_to_reply("Contact:\r\n"); > append_to_reply("Content-Type: > application/sdp\r\n"); > $var(body) = > $(rb{re.subst,/(IP4.).*/\1$var(addr)/g}); >         $var(body) = > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); > t_reply_with_body(200, "OK", > $var(body)); > rtpengine_play_media("call-id=$ci > from-tag=$ft > file=/etc/rtpengine/unk_num.wav"); >         async(sleep(10), after_media); >      } else { >         xlog("######## Entered > route LateMedia3 No SDP received, > Create one from variable > ########\r\n"); >         $var(newbody) = > ("v=0\r\no=Opensips " + $Ts + " 0 > IN IP4 > 10.255.100.147\r\ns=-\r\nc=IN IP4 > 10.255.100.147\r\nt=0 0\r\nm=audio > 3140 RTP/AVP 0 101\r\na > =sendrecv\r\na=rtpmap:0 > PCMU/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 > 0-15\r\n"); > xlog("######################### > Body to RTPENGINE is > ###########################\r\n$var(newbody)\r\n"); > rtpengine_offer("from-tag=$ft > replace-session-connection > trust-address replace-origin > codec-strip-g729",,$var(body),$var(newbody)); > xlog("######################### > Body from RTPENGINE is > ###########################\r\n$var(body)\r\n"); >         $json(reply) := $rtpquery; > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > port); > $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local > address); > append_to_reply("Contact:\r\n"); > append_to_reply("Content-Type: > application/sdp\r\n"); > $var(body) = > $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g}); >         $var(body) = > $(var(body){re.subst,/(audio.)...../\1$var(port)/g}); > xlog("######################### > Body being sent in Reply is > ######################\r\n$var(body)\r\n"); > t_reply_with_body(200, "OK", > $var(body)); > rtpengine_play_media("call-id=$ci > from-tag=$ft > file=/etc/rtpengine/unk_num.wav"); > async(sleep(10), after_media); >         } > } > > route[after_media] >     { if (t_was_cancelled()) { > rtpengine_delete(); >         exit; >     } else { > rtpengine_delete(); > sl_send_reply(486,"Busy here"); >         exit; >     } > } > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From akogan at 5gfuture.com Tue Nov 14 08:01:15 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Tue, 14 Nov 2023 12:01:15 +0400 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> Message-ID: <8ec142cf-00c6-4829-9cbd-59e0f42957a1@5gfuture.com> Hi, The process is one of the UDP workers actually. You can see that error occurred in receive_msg: ERROR:core:receive_msg: no pkg mem left for sip_msg. I also checked that with MI 'ps' command. Usually I got it from the single process. Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com On 13.11.2023 11:50, Bogdan-Andrei Iancu wrote: > Hi Alexander, > > The warning is actually reporting a highly fragmented memory and not a > lack of memory. It reports 303 available Mb, but not a single > continuous slot for like 33K. > > By doing an MI "ps", could you identify the name/type of the process > reporting this issue ? do you get it from a single process or from > multiple processes ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 11/9/23 4:32 PM, Alexander Kogan wrote: >> >> I've tried all of them - Q, F, and HP with the same result. >> >> Best regards, >> Alexander Kogan, >> Director of R&D317724640 >> 5g Future >> http://5gfuture.com >> >> >> On 09.11.2023 18:24, solarmon wrote: >>> What memory management and allocator are you using? >>> >>> I had similar issues using opensips 3.2.7 and the quick workaround >>> was to change from F_MALLOC to HP_MALLOC. But I thought there should >>> have been fixes for such issues in newer versions, so your issue >>> might be different. >>> >>> On Thu, 9 Nov 2023 at 12:33, Alexander Kogan >>> wrote: >>> >>> Hi, >>> >>> of course I did. It doesn't help. >>> >>> Best regards, >>> Alexander Kogan, >>> Director of R&D >>> 5g Future >>> http://5gfuture.com >>> >>> >>> On 09.11.2023 16:03, Marcin Groszek wrote: >>> > You may try adding more pgk memory in >>> > >>> > /etc/sysconfig/opensips i.e. >>> > >>> > P_MEMORY=256 >>> > >>> > >>> > On 11/9/2023 5:07 AM, Alexander Kogan wrote: >>> >> Hello all, >>> >> >>> >> I regularly get a memory error after upgrading to 3.2.14 and >>> 3.2.15 >>> >> builds. >>> >> >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> >> WARNING:core:fm_malloc: not enough contiguous free pkg memory >>> >> (317724640 bytes left, need 33832), attempting >>> defragmentation... >>> >> please increase the "-M" command line parameter! >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> ERROR:core:fm_malloc: >>> >> not enough free pkg memory (317724640 bytes left, need >>> 33832), please >>> >> increase the "-M" command line parameter! >>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>> >> ERROR:core:receive_msg: no pkg mem left for sip_msg >>> >> >>> >> Surprisingly, it happens only with one or two UDP workers of >>> 15 or >>> >> more. I've already created issue >>> >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm >>> waiting for it. >>> >> >>> >> Meanwhile, I'm looking for a way of restarting particular >>> opensips >>> >> process. Is it possible? >>> >> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 14 09:26:03 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Nov 2023 11:26:03 +0200 Subject: [OpenSIPS-Users] Not enough free pkg memory In-Reply-To: <8ec142cf-00c6-4829-9cbd-59e0f42957a1@5gfuture.com> References: <207e9e3a-e7d7-44af-ba1a-5c1364a93869@5gfuture.com> <968471a8-7d5c-6e33-f131-ffbd257d1ddf@voipplus.net> <3ca651c7-fce1-7d00-76f9-2d03cb05d948@opensips.org> <8ec142cf-00c6-4829-9cbd-59e0f42957a1@5gfuture.com> Message-ID: <4c177113-9a21-810a-9218-8f77777ffd9a@opensips.org> Let's continue this on the ticket https://github.com/OpenSIPS/opensips/issues/3235 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/14/23 10:01 AM, Alexander Kogan wrote: > > Hi, > > The process is one of the UDP workers actually. You can see that error > occurred in receive_msg: ERROR:core:receive_msg: no pkg mem left for > sip_msg. I also checked that with MI 'ps' command. Usually I got it > from the single process. > > Best regards, > Alexander Kogan, > Director of R&D > 5g Future > http://5gfuture.com > > > On 13.11.2023 11:50, Bogdan-Andrei Iancu wrote: >> Hi Alexander, >> >> The warning is actually reporting a highly fragmented memory and not >> a lack of memory. It reports 303 available Mb, but not a single >> continuous slot for like 33K. >> >> By doing an MI "ps", could you identify the name/type of the process >> reporting this issue ? do you get it from a single process or from >> multiple processes ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> On 11/9/23 4:32 PM, Alexander Kogan wrote: >>> >>> I've tried all of them - Q, F, and HP with the same result. >>> >>> Best regards, >>> Alexander Kogan, >>> Director of R&D317724640 >>> 5g Future >>> http://5gfuture.com >>> >>> >>> On 09.11.2023 18:24, solarmon wrote: >>>> What memory management and allocator are you using? >>>> >>>> I had similar issues using opensips 3.2.7 and the quick workaround >>>> was to change from F_MALLOC to HP_MALLOC. But I thought there >>>> should have been fixes for such issues in newer versions, so your >>>> issue might be different. >>>> >>>> On Thu, 9 Nov 2023 at 12:33, Alexander Kogan >>>> wrote: >>>> >>>> Hi, >>>> >>>> of course I did. It doesn't help. >>>> >>>> Best regards, >>>> Alexander Kogan, >>>> Director of R&D >>>> 5g Future >>>> http://5gfuture.com >>>> >>>> >>>> On 09.11.2023 16:03, Marcin Groszek wrote: >>>> > You may try adding more pgk memory in >>>> > >>>> > /etc/sysconfig/opensips i.e. >>>> > >>>> > P_MEMORY=256 >>>> > >>>> > >>>> > On 11/9/2023 5:07 AM, Alexander Kogan wrote: >>>> >> Hello all, >>>> >> >>>> >> I regularly get a memory error after upgrading to 3.2.14 and >>>> 3.2.15 >>>> >> builds. >>>> >> >>>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>>> >> WARNING:core:fm_malloc: not enough contiguous free pkg memory >>>> >> (317724640 bytes left, need 33832), attempting >>>> defragmentation... >>>> >> please increase the "-M" command line parameter! >>>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>>> ERROR:core:fm_malloc: >>>> >> not enough free pkg memory (317724640 bytes left, need >>>> 33832), please >>>> >> increase the "-M" command line parameter! >>>> >> Nov  9 11:53:02 FI173 /usr/sbin/opensips[3243]: >>>> >> ERROR:core:receive_msg: no pkg mem left for sip_msg >>>> >> >>>> >> Surprisingly, it happens only with one or two UDP workers of >>>> 15 or >>>> >> more. I've already created issue >>>> >> https://github.com/OpenSIPS/opensips/issues/3235 and I'm >>>> waiting for it. >>>> >> >>>> >> Meanwhile, I'm looking for a way of restarting particular >>>> opensips >>>> >> process. Is it possible? >>>> >> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From ben.bliss at telxl.com Tue Nov 14 14:00:56 2023 From: ben.bliss at telxl.com (Ben Bliss) Date: Tue, 14 Nov 2023 14:00:56 +0000 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable Message-ID: I am using Opensips v.3.2 and I am attempting to replicate the 302 redirect outlined on the blog (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/), but am running into an odd issue. I am using exec to execute an external program, which returns a URI as a variable, which is then used to populate the $branch field, which is then returned as the contact URI in the 302 response. For reasons I cannot explain, even though the URI is correctly formatted in the xlogs which I print off during the call request, when the 302 message is received, the Contact URI is missing the end '>'character. This means the server the 302 is being sent to doesn't ACK the message, as the contact URI is invalid. If I specify the URI directly within OpenSIPs, then the contact header is then formatted correctly. Initially I thought the issue was related to the script returning the variable (bash script), so I swapped to a Golang program, essentially returning the same URI, but the same behaviour occurs with this as well. The script is setup as follows... if ( $si == '10.10.14.71' && is_method("INVITE") ) { exec("/usr/local/bin/sbc-route", $tU, $var(out)); xlog("Value returned from Re-direct Script is $var(out)"); $branch = $var(out); xlog("Target is set as $branch"); send_reply(302, "Server Redirect"); exit; } Can anyone offer any ideas as to why this may be occurring? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 14 14:32:14 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Nov 2023 16:32:14 +0200 Subject: [OpenSIPS-Users] Introducing OpenSIPS 3.5 Message-ID: Dear OpenSIPS'ers, The 3.4 release is still fresh, but we already started to work on planning the next major release, the OpenSIPS 3.5. This release is to be *IMS focused*. Bits and pieces of the*IMS (IP Multimedia Subsystem) *topic were part of the previous OpenSIPS release, but 3.5 aims to fully focus on the IMS part. Considering the traction and need of IMS solutions, we see the implementation of a consistent and large IMS support in OpenSIPS as a mandatory step in order to answer to the needs of the industry. This year we introduce a new concept of an *OpenSIPS Working Group*. And the *IMS OpenSIPS Working Group* is the first one, aiming to gather people with inters in IMS with the goal to draft, design and implement the IMS support in OpenSIPS. More details on this may be found here, please read and act:     https://www.opensips.org/Development/Opensips-3-5-Planning Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Tue Nov 14 14:44:05 2023 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 14 Nov 2023 15:44:05 +0100 Subject: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5 In-Reply-To: References: Message-ID: On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu wrote: > > This year we introduce a new concept of an *OpenSIPS Working Group*. And > the *IMS OpenSIPS Working Group* > is the > first one, aiming to gather people with inters in IMS with the goal to > draft, design and implement the IMS support in OpenSIPS. > > More details on this may be found here, please read and act: > > https://www.opensips.org/Development/Opensips-3-5-Planning > > G R E A T !! (count me on) -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Tue Nov 14 19:51:34 2023 From: Johan at democon.be (Johan De Clercq) Date: Tue, 14 Nov 2023 20:51:34 +0100 Subject: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5 In-Reply-To: References: Message-ID: Me too. On Tue, 14 Nov 2023, 19:02 Giovanni Maruzzelli, wrote: > On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu > wrote: > >> >> This year we introduce a new concept of an *OpenSIPS Working Group*. And >> the *IMS OpenSIPS Working Group* >> is the >> first one, aiming to gather people with inters in IMS with the goal to >> draft, design and implement the IMS support in OpenSIPS. >> >> More details on this may be found here, please read and act: >> >> https://www.opensips.org/Development/Opensips-3-5-Planning >> >> > > > G R E A T !! > (count me on) > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 15 07:15:14 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Nov 2023 09:15:14 +0200 Subject: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5 In-Reply-To: References: Message-ID: Honored to have your support Giovanni :). We will allocate couple of days to (1) start a wiki page with the basic /starting set of requirements and (b) let people subscribe so we can a good pool of brains :) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/14/23 4:44 PM, Giovanni Maruzzelli wrote: > On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu > > wrote: > > > This year we introduce a new concept of an *OpenSIPS Working > Group*. And the *IMS OpenSIPS Working Group* > is > the first one, aiming to gather people with inters in IMS with the > goal to draft, design and implement the IMS support in OpenSIPS. > > More details on this may be found here, please read and act: > > https://www.opensips.org/Development/Opensips-3-5-Planning > > > > > > G R E A T !! > (count me on) > > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 15 07:29:53 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Nov 2023 09:29:53 +0200 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: References: Message-ID: Hi Ben, Could you post the xlog output and the resulting 302 ? Also be sure you are using the latest 3.2 version - please post the `opensips -V` here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/14/23 4:00 PM, Ben Bliss wrote: > > I am using Opensips v.3.2 and I am attempting to replicate the 302 > redirect outlined on the blog > (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/ > ), > but am running into an odd issue. > > I am using exec to execute an external program, which returns a URI as > a variable, which is then used to populate the $branch field, which is > then returned as the contact URI in the 302 response. > > For reasons I cannot explain, even though the URI is correctly > formatted in the xlogs which I print off during the call request, when > the 302 message is received, the Contact URI is missing the end > ‘>‘character. This means the server the 302 is being sent to doesn’t > ACK the message, as the contact URI is invalid. > > If I specify the URI directly within OpenSIPs, then the contact header > is then formatted correctly. > > Initially I thought the issue was related to the script returning the > variable (bash script), so I swapped to a Golang program, essentially > returning the same URI, but the same behaviour occurs with this as well. > > The script is setup as follows… > >         if ( $si == '10.10.14.71' && is_method("INVITE") ) { > > exec("/usr/local/bin/sbc-route", $tU, $var(out)); > >                         xlog("Value returned from Re-direct Script is > $var(out)"); > >                         $branch = $var(out); > >                         xlog("Target is set as $branch"); > >                         send_reply(302, "Server Redirect"); > >                         exit; > >         } > > Can anyone offer any ideas as to why this may be occurring? > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sreeram.narayanan at skit.ai Mon Nov 13 11:26:18 2023 From: sreeram.narayanan at skit.ai (Sreeram Narayanan) Date: Mon, 13 Nov 2023 16:56:18 +0530 Subject: [OpenSIPS-Users] ACK looping behind NAT Message-ID: Hello, I am trying to use OpenSIPs with the load_balancer module to balance inbound calls between 2 Asterisk servers. The setup sits behind a NAT. The OpenSIPs server has a public IP and a private IP. When an INVITE arrives, it can forward it to one of the Asterisk servers and Asterisk responds with a 200 OK. The problem starts when I receive the ACK (from Twilio). The ACK starts bouncing between the public IP and Private IP of the OpenSIPs server. It doesn't reach the Asterisk server and eventually times out. I hope someone can help me with this. Thanks in advance. Here is my configuration: ####### Routing Logic ######## > route { > > if (is_method("INVITE")) { > rtpproxy_engage(); > } > > if ($rm=="INVITE") { > > lb_start_or_next(1,"pstn"); > } > > t_check_trans(); > record_route(); > > t_on_failure("GW_FAILOVER"); > > # route the request > if (!t_relay()) { > sl_reply_error(); > } > > exit; > } > > route[RELAY] { > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > failure_route[GW_FAILOVER] { > if (t_was_cancelled()) { > exit; > } > # failure detection with redirect to next available trunk > if (t_check_status("(408)|([56][0-9][0-9])")) { > xlog("Failed trunk $rd/$du detected \n"); > } > } > -- - Sreeram -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Nov 15 09:16:07 2023 From: nzdealshelp at gmail.com (nz deals) Date: Wed, 15 Nov 2023 22:16:07 +1300 Subject: [OpenSIPS-Users] memcached opensips 3.3 In-Reply-To: References: Message-ID: Hi Bogdan, yes. Thanks On Mon, 13 Nov 2023 at 20:43, Bogdan-Andrei Iancu wrote: > Hi Jason, > > you say the failover (inside the group) works ok when using REDIS, but not > with MemCached ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/3/23 2:04 AM, nz deals wrote: > > After additional testing, I observed that the settings for Redis are > functioning properly, just for your information. It appears there might be > an issue specifically when using memcached in a group. > > Thanks > > Regards, > Jason > > On Thu, 2 Nov 2023 at 21:09, nz deals wrote: > >> Hope everyone is having a good day, >> I've encountered an issue related to making a memcached group. My >> understanding is that when configuring a group, OpenSIPS attempts to >> connect to other memcached servers if one is unresponsive. However, it >> seems to only connect to the first server in the group and perform >> insertions exclusively on that one. In the event that the first server >> becomes inaccessible, I receive an error message stating "Failed to insert: >> CONNECTION FAILURE" and no connection/insertion to the second one. >> >> Here is a snippet of the configuration I'm using: >> modparam("cachedb_memcached", >> "cachedb_url","memcached:main://memcacheserver1:11222,memcachedserver2:11222/") >> >> I've thoroughly reviewed the documentation available, but I couldn't find >> sufficient clarity on this behavior. I would greatly appreciate it if >> someone could provide insights or clarification on this matter. >> >> Thanks >> Jason >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ben.bliss at telxl.com Wed Nov 15 10:19:54 2023 From: ben.bliss at telxl.com (Ben Bliss) Date: Wed, 15 Nov 2023 10:19:54 +0000 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: References: Message-ID: HI Bogdan, Thanks for the reply. I was using 3.2.13, so I have now patched this to 3.2.15, which is the latest version on the 3.2 branch. I repeated the test and got the same behaviour. Please see the below information as requested... Script... if ( $si == '10.10.14.71' && is_method("INVITE") ) { exec("/usr/local/bin/sbc-route", $tU, $var(out)); xlog("Value returned from Re-direct Script is $var(out)"); $branch = $var(out); xlog("Target is set as $branch"); send_reply(302, "Server Redirect"); exit; } Xlog information logged when the call request comes in... 2023-11-15T09:52:55.375039+00:00 uk-rdh-testast2 /usr/sbin/opensips[283656]: Value returned from Re-direct Script is sip:33557719673 at 10.10.4.77 2023-11-15T09:52:55.375456+00:00 uk-rdh-testast2 /usr/sbin/opensips[283656]: Target is set as sip:33557719673 at 10.10.4.77 The 302 response is below, exported from Wireshark... No. Time Source Destination Protocol Length Info 15 09:53:20.012882 10.10.12.70 10.10.14.71 SIP 666 Status: 302 Server Redirect | Frame 15: 666 bytes on wire (5328 bits), 666 bytes captured (5328 bits) Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 User Datagram Protocol, Src Port: 5070, Dst Port: 5060 Session Initiation Protocol (302) Status-Line: SIP/2.0 302 Server Redirect Message Header From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364674-1-1700041999-377836318 To: sip:+4415279xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.6023fbda08f41f7e9ebdc611f91805ef Call-ID: 55197572_103382176 at 151.x.xxx.x [Generated Call-ID: 55197572_103382176 at 151.x.xxx.x] CSeq: 61414 INVITE Via: SIP/2.0/UDP 10.10.14.71:5060;branch=z9hG4bK-6a03d713-a7b27-28f11294-7f0923fc1278 Via: SIP/2.0/UDP 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364674 Via: SIP/2.0/UDP 151.x.xxx.x:5060;branch=z9hG4bK0aB2fefabe20ff62d0e Contact: '); This pushed the closing > from the 'Contact' URI to the "Server" line, as you can see on the outputted 302 below... No. Time Source Destination Protocol Length Info 17 12:12:01.339034 10.10.12.70 10.10.14.71 SIP 667 Status: 302 Server Redirect | Frame 17: 667 bytes on wire (5336 bits), 667 bytes captured (5336 bits) Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 User Datagram Protocol, Src Port: 5070, Dst Port: 5060 Session Initiation Protocol (302) Status-Line: SIP/2.0 302 Server Redirect Message Header From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364667-1-1699963921-901707368 To: sip:080001xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.a1daebfe0f15d4035f73e92edd390da9 Call-ID: 55182316_87520217 at 151.x.xxx.x [Generated Call-ID: 55182316_87520217 at 151.x.xxx.x] CSeq: 706035 INVITE Via: SIP/2.0/UDP 10.10.14.71:5060;branch=z9hG4bK-34260abf-94a28-2449afc3-7f0920d0abc8 Via: SIP/2.0/UDP 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364667 Via: SIP/2.0/UDP 151.x.xxx.x:5060;branch=z9hG4bK0aB220722fa7bfe603d Contact: Server: OpenSIPS (3.2.13 (x86_64/linux)) [Expert Info (Note/Undecoded): Unrecognised SIP header (>server)] [Unrecognised SIP header (>server)] [Severity level: Note] [Group: Undecoded] Content-Length: 0 Whether this is related or not, I cannot say, but thought I would include it in any case. Many Thanks, Ben From: Bogdan-Andrei Iancu Sent: Wednesday, November 15, 2023 7:30 AM To: OpenSIPS users mailling list ; Ben Bliss Subject: Re: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable CAUTION: This email originated from outside your organization. Exercise caution when opening attachments or clicking links, especially from unknown senders. Hi Ben, Could you post the xlog output and the resulting 302 ? Also be sure you are using the latest 3.2 version - please post the `opensips -V` here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/14/23 4:00 PM, Ben Bliss wrote: I am using Opensips v.3.2 and I am attempting to replicate the 302 redirect outlined on the blog (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/), but am running into an odd issue. I am using exec to execute an external program, which returns a URI as a variable, which is then used to populate the $branch field, which is then returned as the contact URI in the 302 response. For reasons I cannot explain, even though the URI is correctly formatted in the xlogs which I print off during the call request, when the 302 message is received, the Contact URI is missing the end '>'character. This means the server the 302 is being sent to doesn't ACK the message, as the contact URI is invalid. If I specify the URI directly within OpenSIPs, then the contact header is then formatted correctly. Initially I thought the issue was related to the script returning the variable (bash script), so I swapped to a Golang program, essentially returning the same URI, but the same behaviour occurs with this as well. The script is setup as follows... if ( $si == '10.10.14.71' && is_method("INVITE") ) { exec("/usr/local/bin/sbc-route", $tU, $var(out)); xlog("Value returned from Re-direct Script is $var(out)"); $branch = $var(out); xlog("Target is set as $branch"); send_reply(302, "Server Redirect"); exit; } Can anyone offer any ideas as to why this may be occurring? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Wed Nov 15 11:00:21 2023 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Wed, 15 Nov 2023 12:00:21 +0100 Subject: [OpenSIPS-Users] dlg_end_dlg in early state, how to rewrite 487 to 404 Message-ID: Hi All, I have a case where I need to terminate an early state dialog but not with a 487 but by a 404. t_check_status() on onreply_route didn’t match, and it’s not allowed to use t_check_status() on local_route what is the best way to do it ? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Nov 15 11:09:50 2023 From: Johan at democon.be (Johan De Clercq) Date: Wed, 15 Nov 2023 12:09:50 +0100 Subject: [OpenSIPS-Users] dlg_end_dlg in early state, how to rewrite 487 to 404 In-Reply-To: References: Message-ID: forward to a self defined route and change it there. route[404to487] { .... } onreply_route { .... route(404to487); } Op wo 15 nov 2023 om 12:04 schreef Alain Bieuzent : > Hi All, > > > > I have a case where I need to terminate an early state dialog but not with > a 487 but by a 404. > > > > t_check_status() on onreply_route didn’t match, and it’s not allowed to > use t_check_status() on local_route > > > > what is the best way to do it ? > > > > thanks > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 15 11:27:28 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Nov 2023 13:27:28 +0200 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: References: Message-ID: <310855c1-1756-d1bb-2283-4e35d89ee7d2@opensips.org> Out of curiosity, if you replace the:     $branch = $var(out); with     $ru = $var(out) is the reply properly formed ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/15/23 12:19 PM, Ben Bliss wrote: > > HI Bogdan, > > Thanks for the reply. > > I was using 3.2.13, so I have now patched this to 3.2.15, which is the > latest version on the 3.2 branch. I repeated the test and got the same > behaviour. Please see the below information as requested… > > Script… > >         if ( $si == '10.10.14.71' && is_method("INVITE") ) { > > exec("/usr/local/bin/sbc-route", $tU, $var(out)); > >                         xlog("Value returned from Re-direct Script is > $var(out)"); > >                         $branch = $var(out); > >                         xlog("Target is set as $branch"); > >                         send_reply(302, "Server Redirect"); > >                         exit; > >         } > > Xlog information logged when the call request comes in… > > 2023-11-15T09:52:55.375039+00:00 uk-rdh-testast2 > /usr/sbin/opensips[283656]: Value returned from Re-direct Script is > sip:33557719673 at 10.10.4.77 > > 2023-11-15T09:52:55.375456+00:00 uk-rdh-testast2 > /usr/sbin/opensips[283656]: Target is set as > sip:33557719673 at 10.10.4.77 > > The 302 response is below, exported from Wireshark… > > No.     Time Source                Destination           Protocol > Length Info > > 15 09:53:20.012882    10.10.12.70 10.10.14.71           SIP      > 666    Status: 302 Server Redirect | > > Frame 15: 666 bytes on wire (5328 bits), 666 bytes captured (5328 bits) > > Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: > aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) > > Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 > > User Datagram Protocol, Src Port: 5070, Dst Port: 5060 > > Session Initiation Protocol (302) > >     Status-Line: SIP/2.0 302 Server Redirect > >     Message Header > >         From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone > ;tag=BN1699364674-1-1700041999-377836318 > >         To: sip:+4415279xxxxx at 149.x.xxx.xx;user=phone > ;tag=5d8f.6023fbda08f41f7e9ebdc611f91805ef > >         Call-ID: 55197572_103382176 at 151.x.xxx.x > > >         [Generated Call-ID: 55197572_103382176 at 151.x.xxx.x > ] > >         CSeq: 61414 INVITE > >         Via: SIP/2.0/UDP > 10.10.14.71:5060;branch=z9hG4bK-6a03d713-a7b27-28f11294-7f0923fc1278 > >         Via: SIP/2.0/UDP > 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364674 > >         Via: SIP/2.0/UDP > 151.x.xxx.x:5060;branch=z9hG4bK0aB2fefabe20ff62d0e > >         Contact: >         [Expert Info (Warning/Malformed): Header has no colon after > the name] > >             [Header has no colon after the name] > >             [Severity level: Warning] > >             [Group: Malformed] > >         Server: OpenSIPS (3.2.15 (x86_64/linux)) > >         Content-Length: 0 > > If it is any help, I did try another method of inputting the Contact > header, which was using the ‘append_to_reply’ command, rather than > using the $branch method, so I just formatted in the URI as such… > > append_to_reply('Contact: <$var(out)>'); > > This pushed the closing > from the ‘Contact’ URI to the “Server” line, > as you can see on the outputted 302 below… > > No.     Time Source                Destination           Protocol > Length Info > >      17 12:12:01.339034 10.10.12.70           10.10.14.71           > SIP      667 Status: 302 Server Redirect | > > Frame 17: 667 bytes on wire (5336 bits), 667 bytes captured (5336 bits) > > Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: > aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) > > Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 > > User Datagram Protocol, Src Port: 5070, Dst Port: 5060 > > Session Initiation Protocol (302) > >     Status-Line: SIP/2.0 302 Server Redirect > >     Message Header > >         From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone > ;tag=BN1699364667-1-1699963921-901707368 > >         To: sip:080001xxxxx at 149.x.xxx.xx;user=phone > ;tag=5d8f.a1daebfe0f15d4035f73e92edd390da9 > >         Call-ID: 55182316_87520217 at 151.x.xxx.x > > >         [Generated Call-ID: 55182316_87520217 at 151.x.xxx.x > ] > >         CSeq: 706035 INVITE > >         Via: SIP/2.0/UDP > 10.10.14.71:5060;branch=z9hG4bK-34260abf-94a28-2449afc3-7f0920d0abc8 > >         Via: SIP/2.0/UDP > 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364667 > >         Via: SIP/2.0/UDP > 151.x.xxx.x:5060;branch=z9hG4bK0aB220722fa7bfe603d > >         Contact: >         >Server: OpenSIPS (3.2.13 (x86_64/linux)) > >             [Expert Info (Note/Undecoded): Unrecognised SIP header > (>server)] > >                 [Unrecognised SIP header (>server)] > >                 [Severity level: Note] > >                 [Group: Undecoded] > >         Content-Length: 0 > > Whether this is related or not, I cannot say, but thought I would > include it in any case. > > Many Thanks, > > Ben > > *From:*Bogdan-Andrei Iancu > *Sent:* Wednesday, November 15, 2023 7:30 AM > *To:* OpenSIPS users mailling list ; Ben > Bliss > *Subject:* Re: [OpenSIPS-Users] Odd URI formatting issue when using > Exec to return a variable > > CAUTION: This email originated from outside your organization. > Exercise caution when opening attachments or clicking links, > especially from unknown senders. > > Hi Ben, > > Could you post the xlog output and the resulting 302 ? Also be sure > you are using the latest 3.2 version - please post the `opensips -V` here. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/14/23 4:00 PM, Ben Bliss wrote: > > I am using Opensips v.3.2 and I am attempting to replicate the 302 > redirect outlined on the blog > (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/ > ), > but am running into an odd issue. > > I am using exec to execute an external program, which returns a > URI as a variable, which is then used to populate the $branch > field, which is then returned as the contact URI in the 302 response. > > For reasons I cannot explain, even though the URI is correctly > formatted in the xlogs which I print off during the call request, > when the 302 message is received, the Contact URI is missing the > end ‘>‘character. This means the server the 302 is being sent to > doesn’t ACK the message, as the contact URI is invalid. > > If I specify the URI directly within OpenSIPs, then the contact > header is then formatted correctly. > > Initially I thought the issue was related to the script returning > the variable (bash script), so I swapped to a Golang program, > essentially returning the same URI, but the same behaviour occurs > with this as well. > > The script is setup as follows… > >         if ( $si == '10.10.14.71' && is_method("INVITE") ) { > > exec("/usr/local/bin/sbc-route", $tU, $var(out)); > >                         xlog("Value returned from Re-direct Script > is $var(out)"); > >                         $branch = $var(out); > >                         xlog("Target is set as $branch"); > >                         send_reply(302, "Server Redirect"); > >                         exit; > >         } > > Can anyone offer any ideas as to why this may be occurring? > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 15 11:32:07 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Nov 2023 13:32:07 +0200 Subject: [OpenSIPS-Users] dlg_end_dlg in early state, how to rewrite 487 to 404 In-Reply-To: References: Message-ID: Hi Alain, IF receiving the 487 reply from callee side, you should end up in the failure route - and there do the t_check_status() and a t_reply(404) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/15/23 1:00 PM, Alain Bieuzent wrote: > > Hi All, > > I have a case where I need to terminate an early state dialog but not > with a 487 but by a 404. > > t_check_status() on onreply_route didn’t match, and it’s not allowed > to use t_check_status() on local_route > > what is the best way to do it ? > > thanks > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 15 11:35:06 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Nov 2023 13:35:06 +0200 Subject: [OpenSIPS-Users] ACK looping behind NAT In-Reply-To: References: Message-ID: <7b1fff87-d0e1-1efa-4b5f-f451d8f161a8@opensips.org> Hi, Ideally you should provide a network capture (pcap) from the OpenSIPS server, covering both incoming and outgoing traffic - this is the only way to understand what is wrong with the call. As attachments are limited to 40K here, consider using some pastebin or other file sharing service. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/13/23 1:26 PM, Sreeram Narayanan via Users wrote: > Hello, > > I am trying to use OpenSIPs with the load_balancer module to balance > inbound calls between 2 Asterisk servers. The setup sits behind a NAT. > The OpenSIPs server has a public IP and a private IP. When an INVITE > arrives, it can forward it to one of the Asterisk servers and Asterisk > responds with a 200 OK. The problem starts when I receive the ACK > (from Twilio). The ACK starts bouncing between the public IP and > Private IP of the OpenSIPs server. It doesn't reach the > Asterisk server and eventually times out. I hope someone can help me > with this. Thanks in advance. > > Here is my configuration: > > ####### Routing Logic ######## > route { > >     if (is_method("INVITE")) { >         rtpproxy_engage(); >     } > >     if ($rm=="INVITE") { > >         lb_start_or_next(1,"pstn"); >     } > >     t_check_trans(); >     record_route(); > >     t_on_failure("GW_FAILOVER"); > >     # route the request >     if (!t_relay()) { >         sl_reply_error(); >     } > >     exit; > } > > route[RELAY] { >     if (!t_relay()) { >         sl_reply_error(); >     } >     exit; > } > > failure_route[GW_FAILOVER] { >     if (t_was_cancelled()) { >         exit; >     } >     # failure detection with redirect to next available trunk >     if (t_check_status("(408)|([56][0-9][0-9])")) { >         xlog("Failed trunk $rd/$du detected \n"); >     } > } > > > -- > - Sreeram > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From greg at switchtel.co.za Thu Nov 16 01:26:36 2023 From: greg at switchtel.co.za (Gregory Massel) Date: Thu, 16 Nov 2023 03:26:36 +0200 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: <310855c1-1756-d1bb-2283-4e35d89ee7d2@opensips.org> References: <310855c1-1756-d1bb-2283-4e35d89ee7d2@opensips.org> Message-ID: <4d369801-102f-4088-9511-8fee094b7863@switchtel.co.za> Just wondering if, perhaps, the script return a line terminated with \r\n instead of just \n ? Or perhaps even just a \n is being pulled into the variable? Perhaps try a {s.trim} transformation? On 2023-11-15 13:27, Bogdan-Andrei Iancu wrote: > Out of curiosity, if you replace the: >     $branch = $var(out); > with >     $ru = $var(out) > is the reply properly formed ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 11/15/23 12:19 PM, Ben Bliss wrote: >> >> HI Bogdan, >> >> Thanks for the reply. >> >> I was using 3.2.13, so I have now patched this to 3.2.15, which is >> the latest version on the 3.2 branch. I repeated the test and got the >> same behaviour. Please see the below information as requested… >> >> Script… >> >>         if ( $si == '10.10.14.71' && is_method("INVITE") ) { >> >> exec("/usr/local/bin/sbc-route", $tU, $var(out)); >> >>                         xlog("Value returned from Re-direct Script is >> $var(out)"); >> >>                         $branch = $var(out); >> >>                         xlog("Target is set as $branch"); >> >>                         send_reply(302, "Server Redirect"); >> >>                         exit; >> >>         } >> >> Xlog information logged when the call request comes in… >> >> 2023-11-15T09:52:55.375039+00:00 uk-rdh-testast2 >> /usr/sbin/opensips[283656]: Value returned from Re-direct Script is >> sip:33557719673 at 10.10.4.77 >> >> 2023-11-15T09:52:55.375456+00:00 uk-rdh-testast2 >> /usr/sbin/opensips[283656]: Target is set as sip:33557719673 at 10.10.4.77 >> >> The 302 response is below, exported from Wireshark… >> >> No.     Time Source                Destination           Protocol >> Length Info >> >> 15 09:53:20.012882 10.10.12.70           10.10.14.71           >> SIP      666 Status: 302 Server Redirect | >> >> Frame 15: 666 bytes on wire (5328 bits), 666 bytes captured (5328 bits) >> >> Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: >> aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) >> >> Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 >> >> User Datagram Protocol, Src Port: 5070, Dst Port: 5060 >> >> Session Initiation Protocol (302) >> >>     Status-Line: SIP/2.0 302 Server Redirect >> >>     Message Header >> >>         From: >> sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364674-1-1700041999-377836318 >> >>         To: >> sip:+4415279xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.6023fbda08f41f7e9ebdc611f91805ef >> >>         Call-ID: 55197572_103382176 at 151.x.xxx.x >> >>         [Generated Call-ID: 55197572_103382176 at 151.x.xxx.x] >> >>         CSeq: 61414 INVITE >> >>         Via: SIP/2.0/UDP >> 10.10.14.71:5060;branch=z9hG4bK-6a03d713-a7b27-28f11294-7f0923fc1278 >> >>         Via: SIP/2.0/UDP >> 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364674 >> >>         Via: SIP/2.0/UDP >> 151.x.xxx.x:5060;branch=z9hG4bK0aB2fefabe20ff62d0e >> >>         Contact: > >>         [Expert Info (Warning/Malformed): Header has no colon after >> the name] >> >>             [Header has no colon after the name] >> >>             [Severity level: Warning] >> >>             [Group: Malformed] >> >>         Server: OpenSIPS (3.2.15 (x86_64/linux)) >> >>         Content-Length: 0 >> >> If it is any help, I did try another method of inputting the Contact >> header, which was using the ‘append_to_reply’ command, rather than >> using the $branch method, so I just formatted in the URI as such… >> >> append_to_reply('Contact: <$var(out)>'); >> >> This pushed the closing > from the ‘Contact’ URI to the “Server” >> line, as you can see on the outputted 302 below… >> >> No.     Time Source                Destination           Protocol >> Length Info >> >>      17 12:12:01.339034 10.10.12.70           10.10.14.71           >> SIP      667 Status: 302 Server Redirect | >> >> Frame 17: 667 bytes on wire (5336 bits), 667 bytes captured (5336 bits) >> >> Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: >> aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) >> >> Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 >> >> User Datagram Protocol, Src Port: 5070, Dst Port: 5060 >> >> Session Initiation Protocol (302) >> >>     Status-Line: SIP/2.0 302 Server Redirect >> >>     Message Header >> >>         From: >> sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364667-1-1699963921-901707368 >> >>         To: >> sip:080001xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.a1daebfe0f15d4035f73e92edd390da9 >> >>         Call-ID: 55182316_87520217 at 151.x.xxx.x >> >>         [Generated Call-ID: 55182316_87520217 at 151.x.xxx.x] >> >>         CSeq: 706035 INVITE >> >>         Via: SIP/2.0/UDP >> 10.10.14.71:5060;branch=z9hG4bK-34260abf-94a28-2449afc3-7f0920d0abc8 >> >>         Via: SIP/2.0/UDP >> 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364667 >> >>         Via: SIP/2.0/UDP >> 151.x.xxx.x:5060;branch=z9hG4bK0aB220722fa7bfe603d >> >>         Contact: > >>         >Server: OpenSIPS (3.2.13 (x86_64/linux)) >> >>             [Expert Info (Note/Undecoded): Unrecognised SIP header >> (>server)] >> >>                 [Unrecognised SIP header (>server)] >> >>                 [Severity level: Note] >> >>                 [Group: Undecoded] >> >>         Content-Length: 0 >> >> Whether this is related or not, I cannot say, but thought I would >> include it in any case. >> >> Many Thanks, >> >> Ben >> >> *From:*Bogdan-Andrei Iancu >> *Sent:* Wednesday, November 15, 2023 7:30 AM >> *To:* OpenSIPS users mailling list ; Ben >> Bliss >> *Subject:* Re: [OpenSIPS-Users] Odd URI formatting issue when using >> Exec to return a variable >> >> CAUTION: This email originated from outside your organization. >> Exercise caution when opening attachments or clicking links, >> especially from unknown senders. >> >> Hi Ben, >> >> Could you post the xlog output and the resulting 302 ? Also be sure >> you are using the latest 3.2 version - please post the `opensips -V` >> here. >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/14/23 4:00 PM, Ben Bliss wrote: >> >> I am using Opensips v.3.2 and I am attempting to replicate the >> 302 redirect outlined on the blog >> (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/), >> but am running into an odd issue. >> >> I am using exec to execute an external program, which returns a >> URI as a variable, which is then used to populate the $branch >> field, which is then returned as the contact URI in the 302 >> response. >> >> For reasons I cannot explain, even though the URI is correctly >> formatted in the xlogs which I print off during the call request, >> when the 302 message is received, the Contact URI is missing the >> end ‘>‘character. This means the server the 302 is being sent to >> doesn’t ACK the message, as the contact URI is invalid. >> >> If I specify the URI directly within OpenSIPs, then the contact >> header is then formatted correctly. >> >> Initially I thought the issue was related to the script returning >> the variable (bash script), so I swapped to a Golang program, >> essentially returning the same URI, but the same behaviour occurs >> with this as well. >> >> The script is setup as follows… >> >>         if ( $si == '10.10.14.71' && is_method("INVITE") ) { >> >> exec("/usr/local/bin/sbc-route", $tU, $var(out)); >> >>                         xlog("Value returned from Re-direct >> Script is $var(out)"); >> >>                         $branch = $var(out); >> >>                         xlog("Target is set as $branch"); >> >>                         send_reply(302, "Server Redirect"); >> >>                         exit; >> >>         } >> >> Can anyone offer any ideas as to why this may be occurring? >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 16 12:55:58 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2023 14:55:58 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location Message-ID: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> Dear OpenSIPS'ers, Thanks to the support and effort of the community, we have some really good nominee cities for OpenSIPS Summit 2024. So here is the list of validated cities:     https://bit.ly/summit-2024-location-poll Please help us with the final decision of the event location. Your opinion matters to us, so please help us to build a great event for you. Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com From social at bohboh.info Thu Nov 16 13:13:08 2023 From: social at bohboh.info (Social Boh) Date: Thu, 16 Nov 2023 08:13:08 -0500 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location In-Reply-To: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> References: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> Message-ID: <32927554-ba8e-4028-ac72-821838b76219@bohboh.info> I can't vote without a account or register... maybe would be better vote without login. Regards --- I'm SoCIaL, MayBe El 16/11/2023 a las 7:55 a. m., Bogdan-Andrei Iancu escribió: > Dear OpenSIPS'ers, > > Thanks to the support and effort of the community, we have some really > good nominee cities for OpenSIPS Summit 2024. > > So here is the list of validated cities: >     https://bit.ly/summit-2024-location-poll > > Please help us with the final decision of the event location. Your > opinion matters to us, so please help us to build a great event for you. > > Best regards, > From bogdan at opensips.org Thu Nov 16 13:16:47 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2023 15:16:47 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location In-Reply-To: <32927554-ba8e-4028-ac72-821838b76219@bohboh.info> References: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> <32927554-ba8e-4028-ac72-821838b76219@bohboh.info> Message-ID: In terms of easiness, I agree, but in terms of fairness, I do not :) - we need a way to guarantee one vote per person, otherwise the whole process will be irrelevant. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/16/23 3:13 PM, Social Boh wrote: > I can't vote without a account or register... maybe would be better > vote without login. > > Regards > > --- > I'm SoCIaL, MayBe > > El 16/11/2023 a las 7:55 a. m., Bogdan-Andrei Iancu escribió: >> Dear OpenSIPS'ers, >> >> Thanks to the support and effort of the community, we have some >> really good nominee cities for OpenSIPS Summit 2024. >> >> So here is the list of validated cities: >>     https://bit.ly/summit-2024-location-poll >> >> Please help us with the final decision of the event location. Your >> opinion matters to us, so please help us to build a great event for you. >> >> Best regards, >> From alberto.rinaudo at gmail.com Thu Nov 16 13:48:30 2023 From: alberto.rinaudo at gmail.com (Alberto) Date: Thu, 16 Nov 2023 13:48:30 +0000 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location In-Reply-To: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> References: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> Message-ID: Was reading through and just wanted to mention that Trieste does have an airport, it's small, but very well connected https://maps.app.goo.gl/Rz47zTNWVoZuhi248 [image: image.png] On Thu, 16 Nov 2023 at 13:02, Bogdan-Andrei Iancu wrote: > Dear OpenSIPS'ers, > > Thanks to the support and effort of the community, we have some really > good nominee cities for OpenSIPS Summit 2024. > > So here is the list of validated cities: > https://bit.ly/summit-2024-location-poll > > Please help us with the final decision of the event location. Your > opinion matters to us, so please help us to build a great event for you. > > Best regards, > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 112638 bytes Desc: not available URL: From bogdan at opensips.org Thu Nov 16 15:05:11 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Nov 2023 17:05:11 +0200 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: <4d369801-102f-4088-9511-8fee094b7863@switchtel.co.za> References: <310855c1-1756-d1bb-2283-4e35d89ee7d2@opensips.org> <4d369801-102f-4088-9511-8fee094b7863@switchtel.co.za> Message-ID: I agree here, it looks like the exec output has some trailing chars, like an \n ..... Ben, try doing     $branch = $(var(out){s.trim}); See https://opensips.org/Documentation/Script-Tran-3-4#s.trim Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/16/23 3:26 AM, Gregory Massel via Users wrote: > > Just wondering if, perhaps, the script return a line terminated with > \r\n instead of just \n ? Or perhaps even just a \n is being pulled > into the variable? > > Perhaps try a {s.trim} transformation? > > On 2023-11-15 13:27, Bogdan-Andrei Iancu wrote: >> Out of curiosity, if you replace the: >>     $branch = $var(out); >> with >>     $ru = $var(out) >> is the reply properly formed ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> On 11/15/23 12:19 PM, Ben Bliss wrote: >>> >>> HI Bogdan, >>> >>> Thanks for the reply. >>> >>> I was using 3.2.13, so I have now patched this to 3.2.15, which is >>> the latest version on the 3.2 branch. I repeated the test and got >>> the same behaviour. Please see the below information as requested… >>> >>> Script… >>> >>>         if ( $si == '10.10.14.71' && is_method("INVITE") ) { >>> >>> exec("/usr/local/bin/sbc-route", $tU, $var(out)); >>> >>>                         xlog("Value returned from Re-direct Script >>> is $var(out)"); >>> >>>                         $branch = $var(out); >>> >>>                         xlog("Target is set as $branch"); >>> >>>                         send_reply(302, "Server Redirect"); >>> >>>                         exit; >>> >>>         } >>> >>> Xlog information logged when the call request comes in… >>> >>> 2023-11-15T09:52:55.375039+00:00 uk-rdh-testast2 >>> /usr/sbin/opensips[283656]: Value returned from Re-direct Script is >>> sip:33557719673 at 10.10.4.77 >>> >>> 2023-11-15T09:52:55.375456+00:00 uk-rdh-testast2 >>> /usr/sbin/opensips[283656]: Target is set as sip:33557719673 at 10.10.4.77 >>> >>> The 302 response is below, exported from Wireshark… >>> >>> No.     Time Source                Destination           Protocol >>> Length Info >>> >>> 15 09:53:20.012882 10.10.12.70           10.10.14.71           SIP >>> 666    Status: 302 Server Redirect | >>> >>> Frame 15: 666 bytes on wire (5328 bits), 666 bytes captured (5328 bits) >>> >>> Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: >>> aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) >>> >>> Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 >>> >>> User Datagram Protocol, Src Port: 5070, Dst Port: 5060 >>> >>> Session Initiation Protocol (302) >>> >>>     Status-Line: SIP/2.0 302 Server Redirect >>> >>>     Message Header >>> >>>         From: >>> sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364674-1-1700041999-377836318 >>> >>>         To: >>> sip:+4415279xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.6023fbda08f41f7e9ebdc611f91805ef >>> >>>         Call-ID: 55197572_103382176 at 151.x.xxx.x >>> >>>         [Generated Call-ID: 55197572_103382176 at 151.x.xxx.x] >>> >>>         CSeq: 61414 INVITE >>> >>>         Via: SIP/2.0/UDP >>> 10.10.14.71:5060;branch=z9hG4bK-6a03d713-a7b27-28f11294-7f0923fc1278 >>> >>>         Via: SIP/2.0/UDP >>> 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364674 >>> >>>         Via: SIP/2.0/UDP >>> 151.x.xxx.x:5060;branch=z9hG4bK0aB2fefabe20ff62d0e >>> >>>         Contact: >> >>>         [Expert Info (Warning/Malformed): Header has no colon after >>> the name] >>> >>>             [Header has no colon after the name] >>> >>>             [Severity level: Warning] >>> >>>             [Group: Malformed] >>> >>>         Server: OpenSIPS (3.2.15 (x86_64/linux)) >>> >>>         Content-Length: 0 >>> >>> If it is any help, I did try another method of inputting the Contact >>> header, which was using the ‘append_to_reply’ command, rather than >>> using the $branch method, so I just formatted in the URI as such… >>> >>> append_to_reply('Contact: <$var(out)>'); >>> >>> This pushed the closing > from the ‘Contact’ URI to the “Server” >>> line, as you can see on the outputted 302 below… >>> >>> No.     Time Source                Destination           Protocol >>> Length Info >>> >>>      17 12:12:01.339034 10.10.12.70           10.10.14.71           >>> SIP 667    Status: 302 Server Redirect | >>> >>> Frame 17: 667 bytes on wire (5336 bits), 667 bytes captured (5336 bits) >>> >>> Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: >>> aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) >>> >>> Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 >>> >>> User Datagram Protocol, Src Port: 5070, Dst Port: 5060 >>> >>> Session Initiation Protocol (302) >>> >>>     Status-Line: SIP/2.0 302 Server Redirect >>> >>>     Message Header >>> >>>         From: >>> sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364667-1-1699963921-901707368 >>> >>>         To: >>> sip:080001xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.a1daebfe0f15d4035f73e92edd390da9 >>> >>>         Call-ID: 55182316_87520217 at 151.x.xxx.x >>> >>>         [Generated Call-ID: 55182316_87520217 at 151.x.xxx.x] >>> >>>         CSeq: 706035 INVITE >>> >>>         Via: SIP/2.0/UDP >>> 10.10.14.71:5060;branch=z9hG4bK-34260abf-94a28-2449afc3-7f0920d0abc8 >>> >>>         Via: SIP/2.0/UDP >>> 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364667 >>> >>>         Via: SIP/2.0/UDP >>> 151.x.xxx.x:5060;branch=z9hG4bK0aB220722fa7bfe603d >>> >>>         Contact: >> >>>         >Server: OpenSIPS (3.2.13 (x86_64/linux)) >>> >>>             [Expert Info (Note/Undecoded): Unrecognised SIP header >>> (>server)] >>> >>>                 [Unrecognised SIP header (>server)] >>> >>>                 [Severity level: Note] >>> >>>                 [Group: Undecoded] >>> >>>         Content-Length: 0 >>> >>> Whether this is related or not, I cannot say, but thought I would >>> include it in any case. >>> >>> Many Thanks, >>> >>> Ben >>> >>> *From:*Bogdan-Andrei Iancu >>> *Sent:* Wednesday, November 15, 2023 7:30 AM >>> *To:* OpenSIPS users mailling list ; Ben >>> Bliss >>> *Subject:* Re: [OpenSIPS-Users] Odd URI formatting issue when using >>> Exec to return a variable >>> >>> CAUTION: This email originated from outside your organization. >>> Exercise caution when opening attachments or clicking links, >>> especially from unknown senders. >>> >>> Hi Ben, >>> >>> Could you post the xlog output and the resulting 302 ? Also be sure >>> you are using the latest 3.2 version - please post the `opensips -V` >>> here. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/14/23 4:00 PM, Ben Bliss wrote: >>> >>> I am using Opensips v.3.2 and I am attempting to replicate the >>> 302 redirect outlined on the blog >>> (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/), >>> but am running into an odd issue. >>> >>> I am using exec to execute an external program, which returns a >>> URI as a variable, which is then used to populate the $branch >>> field, which is then returned as the contact URI in the 302 >>> response. >>> >>> For reasons I cannot explain, even though the URI is correctly >>> formatted in the xlogs which I print off during the call >>> request, when the 302 message is received, the Contact URI is >>> missing the end ‘>‘character. This means the server the 302 is >>> being sent to doesn’t ACK the message, as the contact URI is >>> invalid. >>> >>> If I specify the URI directly within OpenSIPs, then the contact >>> header is then formatted correctly. >>> >>> Initially I thought the issue was related to the script >>> returning the variable (bash script), so I swapped to a Golang >>> program, essentially returning the same URI, but the same >>> behaviour occurs with this as well. >>> >>> The script is setup as follows… >>> >>>         if ( $si == '10.10.14.71' && is_method("INVITE") ) { >>> >>> exec("/usr/local/bin/sbc-route", $tU, $var(out)); >>> >>>                         xlog("Value returned from Re-direct >>> Script is $var(out)"); >>> >>>                         $branch = $var(out); >>> >>>                         xlog("Target is set as $branch"); >>> >>> send_reply(302, "Server Redirect"); >>> >>>                         exit; >>> >>>         } >>> >>> Can anyone offer any ideas as to why this may be occurring? >>> >>> >>> >>> _______________________________________________ >>> >>> Users mailing list >>> >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Mon Nov 20 17:27:15 2023 From: rob.dyck at telus.net (Robert Dyck) Date: Mon, 20 Nov 2023 09:27:15 -0800 Subject: [OpenSIPS-Users] Learning about resource lists Message-ID: <46085560.fMDQidcC6G@leno.mylan> The context here is subscription to presence by way of a resource list. The learning curve is steep. I have read the tutorial. The tutorial gives an example of a rls-service xml document. In the example the resource list is contained within the services document. Various other examples I have found use a separate document to hold the list. The services document then references the list document. *https://xcap.example.com/xcap-root/resource-lists/users*/ sip:alice at example.com/index/~~/resource-lists/list%5b at name=%22l1%22%5d If I use an integrated server the xml documents reside in a local database rather than the file system. Http isn't going to work. How would one reference the database and table using rls-services document? Or is a separate resource-lists document not supported when using an integrated rls server? -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Mon Nov 20 21:11:36 2023 From: ag at ag-projects.com (Adrian Georgescu) Date: Mon, 20 Nov 2023 18:11:36 -0300 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: <46085560.fMDQidcC6G@leno.mylan> References: <46085560.fMDQidcC6G@leno.mylan> Message-ID: <799FAE93-340B-497D-81B9-2F5E54EC5B54@ag-projects.com> XCAP is a failure. Not that we did not try, it was a bad idea and it failed. — Adrian > On 20 Nov 2023, at 14:27, Robert Dyck wrote: > > The context here is subscription to presence by way of a resource list. The learning curve is steep. I have read the tutorial. The tutorial gives an example of a rls-service xml document. In the example the resource list is contained within the services document. Various other examples I have found use a separate document to hold the list. The services document then references the list document. > > https://xcap.example.com/xcap-root/resource-lists/users/sip:alice at example.com/index/~~/resource-lists/list%5b at name=%22l1%22%5d > If I use an integrated server the xml documents reside in a local database rather than the file system. Http isn't going to work. How would one reference the database and table using rls-services document? Or is a separate resource-lists document not supported when using an integrated rls server? > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Nov 22 05:04:27 2023 From: nzdealshelp at gmail.com (nz deals) Date: Wed, 22 Nov 2023 18:04:27 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue Message-ID: Hi folks, Is there any bug in the CANCEL handling in the version 3.3.x? I have a weird issue, The INVITE have a VIA header as my private ip Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX But when i CANCEL the call, the CANCEL have my public ip in the VIA header. Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx Anyway can I change the via in the cancel? Regards, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 22 10:02:26 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2023 12:02:26 +0200 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: <799FAE93-340B-497D-81B9-2F5E54EC5B54@ag-projects.com> References: <46085560.fMDQidcC6G@leno.mylan> <799FAE93-340B-497D-81B9-2F5E54EC5B54@ag-projects.com> Message-ID: <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> HI Adrian, should we understand the everything related to xcap, like RLS, buddy list, auth, etc are dropped dead at this time? if so, are you aware of any replacement  / alternatives here ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/20/23 11:11 PM, Adrian Georgescu wrote: > XCAP is a failure. Not that we did not try, it was a bad idea and it > failed. > > — > Adrian > > > > >> On 20 Nov 2023, at 14:27, Robert Dyck wrote: >> >> The context here is subscription to presence by way of a resource >> list. The learning curve is steep. I have read the tutorial. The >> tutorial gives an example of a rls-service xml document. In the >> example the resource list is contained within the services document. >> Various other examples I have found use a separate document to hold >> the list. The services document then references the list document. >> >> *https://xcap.example.com/xcap-root/resource-lists/users*/sip:alice at example.com/index/~~/resource-lists/list%5b at name=%22l1%22%5d >> If I use an integrated server the xml documents reside in a local >> database rather than the file system. Http isn't going to work. How >> would one reference the database and table using rls-services >> document? Or is a separate resource-lists document not supported when >> using an integrated rls server? >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 22 10:07:33 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2023 12:07:33 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: Message-ID: Hi Jason, The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due to an internal timeout / forking? ALso do you use any advertising in your setup? if yes, is it per socket, global or ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/22/23 7:04 AM, nz deals wrote: > Hi folks, > > Is there any bug in the CANCEL handling in the version 3.3.x? > I have a weird issue, > The INVITE have a VIA header as my private ip > Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX > > But when i CANCEL the call, the CANCEL have my public ip in the VIA > header. > Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx > > Anyway can I change the via in the cancel? > > Regards, > Jason > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Wed Nov 22 10:24:52 2023 From: nzdealshelp at gmail.com (nz deals) Date: Wed, 22 Nov 2023 23:24:52 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: Message-ID: Thanks Bogdan, It is due to receiving CANCEL from the caller. Yes using an advertised_address globally. One of our public addresses. Thanks On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu wrote: > Hi Jason, > > The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due to > an internal timeout / forking? ALso do you use any advertising in your > setup? if yes, is it per socket, global or ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 7:04 AM, nz deals wrote: > > Hi folks, > > Is there any bug in the CANCEL handling in the version 3.3.x? > I have a weird issue, > The INVITE have a VIA header as my private ip > Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX > > But when i CANCEL the call, the CANCEL have my public ip in the VIA header. > Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx > > Anyway can I change the via in the cancel? > > Regards, > Jason > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 22 11:58:31 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2023 13:58:31 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location In-Reply-To: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> References: <61c25c3b-4da8-6724-0097-70d4a06fe91c@opensips.org> Message-ID: <3a7c7626-b907-51b8-3020-35decbe33b1b@opensips.org> The people has spoken, *Valencia* will be the city to host OpenSIPS Summit 2024: This is possible thanks to the great help provided by the QXIP team, as our local organizer. Now we have to work out the details, we will update ASAP. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/16/23 2:55 PM, Bogdan-Andrei Iancu wrote: > Dear OpenSIPS'ers, > > Thanks to the support and effort of the community, we have some really > good nominee cities for OpenSIPS Summit 2024. > > So here is the list of validated cities: >     https://bit.ly/summit-2024-location-poll > > Please help us with the final decision of the event location. Your > opinion matters to us, so please help us to build a great event for you. > > Best regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Wed Nov 22 12:01:30 2023 From: ag at ag-projects.com (Adrian Georgescu) Date: Wed, 22 Nov 2023 09:01:30 -0300 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> References: <46085560.fMDQidcC6G@leno.mylan> <799FAE93-340B-497D-81B9-2F5E54EC5B54@ag-projects.com> <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> Message-ID: Hi Bogdan, My two cents. The reality is that adoption of XCAP is practically zero. Even if you build a client, you cannot make it interoperable with another, and XCAP was suppose to be interoperable. If I build a buddy list on one client and I cannot load it in another client, it makes no sense. I think that anyone building a SIP app that needs to store/fetch data on the SIP server can better do it using PUT/GET with a JSON, for example we took this path for Sylk client rather than implementing XCAP again. This is not interoperable between different clients, but there is no replacement standard for XCAP either and is much cheaper and more reliable to do it like this. As far as OpenSIPS is concerned one can probably make a new module or better, modify that existing RLS module so that it can read contacts directly from a database table with a schema that can be defined by the user. In the end what one needs is a list of URIs and a flag to see who is granted to see your presence, is a very simple database model. — Adrian > On 22 Nov 2023, at 07:02, Bogdan-Andrei Iancu wrote: > > HI Adrian, > > should we understand the everything related to xcap, like RLS, buddy list, auth, etc are dropped dead at this time? if so, are you aware of any replacement / alternatives here ? > > Thanks and regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > On 11/20/23 11:11 PM, Adrian Georgescu wrote: >> XCAP is a failure. Not that we did not try, it was a bad idea and it failed. >> >> — >> Adrian >> >> >> >> >>> On 20 Nov 2023, at 14:27, Robert Dyck wrote: >>> >>> The context here is subscription to presence by way of a resource list. The learning curve is steep. I have read the tutorial. The tutorial gives an example of a rls-service xml document. In the example the resource list is contained within the services document. Various other examples I have found use a separate document to hold the list. The services document then references the list document. >>> >>> https://xcap.example.com/xcap-root/resource-lists/users/sip:alice at example.com/index/~~/resource-lists/list%5b at name=%22l1%22%5d >>> If I use an integrated server the xml documents reside in a local database rather than the file system. Http isn't going to work. How would one reference the database and table using rls-services document? Or is a separate resource-lists document not supported when using an integrated rls server? >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 22 12:49:18 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Nov 2023 14:49:18 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: Message-ID: What is your exact opensips version ( `opensips -V` ) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/22/23 12:24 PM, nz deals wrote: > Thanks Bogdan, > > It is due to receiving CANCEL from the caller. Yes using an > advertised_address globally. One of our public addresses. > > Thanks > > On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu > wrote: > > Hi Jason, > > The CANCEL generated by OpenSIPS, is it due to a received CANCEL > or due to an internal timeout / forking? ALso do you use any > advertising in your setup? if yes, is it per socket, global or ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 7:04 AM, nz deals wrote: >> Hi folks, >> >> Is there any bug in the CANCEL handling in the version 3.3.x? >> I have a weird issue, >> The INVITE have a VIA header as my private ip >> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >> >> But when i CANCEL the call, the CANCEL have my public ip in the >> VIA header. >> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >> >> Anyway can I change the via in the cancel? >> >> Regards, >> Jason >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Wed Nov 22 14:57:16 2023 From: rob.dyck at telus.net (Robert Dyck) Date: Wed, 22 Nov 2023 06:57:16 -0800 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: References: <46085560.fMDQidcC6G@leno.mylan> <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> Message-ID: <2779211.BddDVKsqQX@leno.mylan> Is the RLS tutorial valid? Do we know that there are working examples? On Wednesday, November 22, 2023 4:01:30 A.M. PST Adrian Georgescu wrote: > Hi Bogdan, > > My two cents. The reality is that adoption of XCAP is practically zero. Even > if you build a client, you cannot make it interoperable with another, and > XCAP was suppose to be interoperable. If I build a buddy list on one client > and I cannot load it in another client, it makes no sense. > > I think that anyone building a SIP app that needs to store/fetch data on the > SIP server can better do it using PUT/GET with a JSON, for example we took > this path for Sylk client rather than implementing XCAP again. This is not > interoperable between different clients, but there is no replacement > standard for XCAP either and is much cheaper and more reliable to do it > like this. > > As far as OpenSIPS is concerned one can probably make a new module or > better, modify that existing RLS module so that it can read contacts > directly from a database table with a schema that can be defined by the > user. In the end what one needs is a list of URIs and a flag to see who is > granted to see your presence, is a very simple database model. > > — > Adrian > > > On 22 Nov 2023, at 07:02, Bogdan-Andrei Iancu wrote: > > > > HI Adrian, > > > > should we understand the everything related to xcap, like RLS, buddy list, > > auth, etc are dropped dead at this time? if so, are you aware of any > > replacement / alternatives here ? > > > > Thanks and regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > > > https://www.opensips-solutions.com > > https://www.siphub.com > > > > On 11/20/23 11:11 PM, Adrian Georgescu wrote: > >> XCAP is a failure. Not that we did not try, it was a bad idea and it > >> failed. > >> > >> — > >> Adrian > >> > >>> On 20 Nov 2023, at 14:27, Robert Dyck > >>> wrote: > >>> > >>> The context here is subscription to presence by way of a resource list. > >>> The learning curve is steep. I have read the tutorial. The tutorial > >>> gives an example of a rls-service xml document. In the example the > >>> resource list is contained within the services document. Various other > >>> examples I have found use a separate document to hold the list. The > >>> services document then references the list document. > >>> > >>> https://xcap.example.com/xcap-root/resource-lists/users/s > >>> ip:alice at example.com/index/~~/resource-lists/list%5b at name=%22l1%22%5d >>> esource-list> If I use an integrated server the xml documents reside in > >>> a local database rather than the file system. Http isn't going to work. > >>> How would one reference the database and table using rls-services > >>> document? Or is a separate resource-lists document not supported when > >>> using an integrated rls server? > >>> _______________________________________________ > >>> Users mailing list > >>> Users at lists.opensips.org > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >> > >> _______________________________________________ > >> Users mailing list > >> Users at lists.opensips.org > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From ag at ag-projects.com Wed Nov 22 15:05:15 2023 From: ag at ag-projects.com (Adrian Georgescu) Date: Wed, 22 Nov 2023 12:05:15 -0300 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: <2779211.BddDVKsqQX@leno.mylan> References: <46085560.fMDQidcC6G@leno.mylan> <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> <2779211.BddDVKsqQX@leno.mylan> Message-ID: <0A715681-5AE6-4B07-8276-A205FF706EF0@ag-projects.com> You can try these command line scripts: https://sipsimpleclient.org/testing/ Presence • sip-publish-presence - PUBLISH presence to a Presence Agent • sip-subscribe-winfo - SUBSCRIBE to the watcher list for given SIP address on the Presence Agent • sip-subcribe-presence - SUBSCRIBE to Presence Event for a given SIP address • sip-subscribe-rls - SUBSCRIBE for Presence Event to a list managed by a Resource List Server • sip-subscribe-xcap-diff - SUBSCRIBE for xcap-diff Event to monitor changes to XCAP documents • sip-subscribe-mwi - SUBSCRIBE for Message Waiting Indicator — Adrian > On 22 Nov 2023, at 11:57, Robert Dyck wrote: > > Is the RLS tutorial valid? Do we know that there are working examples? From nzdealshelp at gmail.com Wed Nov 22 20:47:15 2023 From: nzdealshelp at gmail.com (nz deals) Date: Thu, 23 Nov 2023 09:47:15 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: Message-ID: Hi Bogdan, My opensips version is opensips 3.3.5 (x86_64/linux) Regards, Jason On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu wrote: > What is your exact opensips version ( `opensips -V` ) ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 12:24 PM, nz deals wrote: > > Thanks Bogdan, > > It is due to receiving CANCEL from the caller. Yes using an > advertised_address globally. One of our public addresses. > > Thanks > > On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu > wrote: > >> Hi Jason, >> >> The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due >> to an internal timeout / forking? ALso do you use any advertising in your >> setup? if yes, is it per socket, global or ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 7:04 AM, nz deals wrote: >> >> Hi folks, >> >> Is there any bug in the CANCEL handling in the version 3.3.x? >> I have a weird issue, >> The INVITE have a VIA header as my private ip >> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >> >> But when i CANCEL the call, the CANCEL have my public ip in the VIA >> header. >> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >> >> Anyway can I change the via in the cancel? >> >> Regards, >> Jason >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at gmail.com Thu Nov 23 11:56:50 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Thu, 23 Nov 2023 12:56:50 +0100 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: Hello! The script that I'm trying to convert to B2B uses the dispatcher module to balance traffic and accordingly, I have a question: If I call *ds_select_dst()* before jumping into the b2b context, then how can I catch the request in *failure_route*, pull out the next peer through *ds_next_dst()*, and return it to b2b again? чт, 19 окт. 2023 г. в 21:55, Ovidiu Sas : > If the internal has a private ip and the external has a public ip, then it > should work ok. > > On Thu, Oct 19, 2023 at 03:14 Denys Pozniak > wrote: > >> > If you have one socket per transport, then the automatic selection >> should work ok. >> For example, I have 3 external interfaces (sip/udp, sip/tcp, sip/tls) and >> 1 internal one (sip/udp). >> If a request comes for an internal one, how will the outgoing routing >> proceed as per needed transport (eg, this is sip/tls) in b2b? What will >> OpenSIPS look at ($ru/Route/...)? >> >> ср, 18 окт. 2023 г. в 19:34, Ovidiu Sas : >> >>> I haven’t experimented with this … so I can’t comment. >>> >>> -ovidiu >>> >>> On Wed, Oct 18, 2023 at 11:13 Denys Pozniak >>> wrote: >>> >>>> and comment please this question if possible: >>>> - If nodes operate in anycast + clusterer mode, then in what route >>>> should function t_anycast_replicate() be called? >>>> >>>> >>>> ср, 18 окт. 2023 г. в 16:49, Denys Pozniak : >>>> >>>>> Thank you for such a cool and detailed answer! >>>>> But I would like to clarify the following: >>>>> >>>>> >Custom headers can be set for each b2b_entity during the entity setup >>>>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>>>> Do I understand correctly that this only works for initial invites? >>>>> >>>>> >For now, the outgoing socket is set by opensips internally. There were >>>>> some fixes related to this, make sure that you are using the latest >>>>> opensips version. >>>>> But I have several sockets and how can I influence the choice? >>>>> >>>>> ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : >>>>> >>>>>> Hello Denys, >>>>>> >>>>>> Dialog module should *never* be mixed with the b2b module. >>>>>> For b2b calls one can use the $b2b_logic.ctx vars [1]. >>>>>> For b2b tracing use the 'b' scope in trace() [2]. >>>>>> There are some issues as not all provisional replies are traced [3]. >>>>>> Custom headers can be set for each b2b_entity during the entity setup >>>>>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>>>>> For now, the outgoing socket is set by opensips internally. There were >>>>>> some fixes related to this, make sure that you are using the latest >>>>>> opensips version. >>>>>> >>>>>> -ovidiu >>>>>> >>>>>> [1] >>>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >>>>>> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >>>>>> [3] https://github.com/OpenSIPS/opensips/issues/3194 >>>>>> [4] >>>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >>>>>> [5] >>>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >>>>>> >>>>>> >>>>>> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak < >>>>>> denys.pozniak at gmail.com> wrote: >>>>>> > >>>>>> > Hello! >>>>>> > >>>>>> > I'm trying to implement topology hiding (b2b_logic) in an existing >>>>>> config, but there are points that are not clear to me: >>>>>> > - Can I somehow use dialog variables in b2b mode? if not, any >>>>>> alternative? I see they are created for the initial leg (with state=3), but >>>>>> not available in the b2b mode; >>>>>> > - How to configure a tracer (proto_hep) in b2b mode? >>>>>> > - How to convert transport and select outgoing socket (the question >>>>>> has already been asked in a separate mail)? >>>>>> > - How can I add the necessary custom X-headers also for in-dialog >>>>>> requests (for example, for re-INVITE)? It does not work via insert_hf in >>>>>> [script_req_route] neither via function b2b_client_new() as well; >>>>>> > - If nodes operate in anycast + clusterer mode, then in what route >>>>>> should function t_anycast_replicate() be called? >>>>>> > >>>>>> > -- >>>>>> > >>>>>> > BR, >>>>>> > Denys Pozniak >>>>>> > >>>>>> > >>>>>> > _______________________________________________ >>>>>> > Users mailing list >>>>>> > Users at lists.opensips.org >>>>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> VoIP Embedded, Inc. >>>>>> http://www.voipembedded.com >>>>>> >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users at lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> BR, >>>>> Denys Pozniak >>>>> >>>>> >>>>> >>>> >>>> -- >>>> >>>> BR, >>>> Denys Pozniak >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> >> BR, >> Denys Pozniak >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From ben.bliss at telxl.com Fri Nov 17 08:03:23 2023 From: ben.bliss at telxl.com (Ben Bliss) Date: Fri, 17 Nov 2023 08:03:23 +0000 Subject: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable In-Reply-To: References: <310855c1-1756-d1bb-2283-4e35d89ee7d2@opensips.org> <4d369801-102f-4088-9511-8fee094b7863@switchtel.co.za> Message-ID: Bogdan, Thank you for your suggestions. That has sorted it, the URI is now coming back in the correct format, and the re-direct is now working. Many thanks! Ben From: Bogdan-Andrei Iancu Sent: Thursday, November 16, 2023 3:05 PM To: Gregory Massel ; OpenSIPS users mailling list ; Ben Bliss Subject: Re: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable CAUTION: This email originated from outside your organization. Exercise caution when opening attachments or clicking links, especially from unknown senders. I agree here, it looks like the exec output has some trailing chars, like an \n ..... Ben, try doing $branch = $(var(out){s.trim}); See https://opensips.org/Documentation/Script-Tran-3-4#s.trim Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/16/23 3:26 AM, Gregory Massel via Users wrote: Just wondering if, perhaps, the script return a line terminated with \r\n instead of just \n ? Or perhaps even just a \n is being pulled into the variable? Perhaps try a {s.trim} transformation? On 2023-11-15 13:27, Bogdan-Andrei Iancu wrote: Out of curiosity, if you replace the: $branch = $var(out); with $ru = $var(out) is the reply properly formed ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/15/23 12:19 PM, Ben Bliss wrote: HI Bogdan, Thanks for the reply. I was using 3.2.13, so I have now patched this to 3.2.15, which is the latest version on the 3.2 branch. I repeated the test and got the same behaviour. Please see the below information as requested… Script… if ( $si == '10.10.14.71' && is_method("INVITE") ) { exec("/usr/local/bin/sbc-route", $tU, $var(out)); xlog("Value returned from Re-direct Script is $var(out)"); $branch = $var(out); xlog("Target is set as $branch"); send_reply(302, "Server Redirect"); exit; } Xlog information logged when the call request comes in… 2023-11-15T09:52:55.375039+00:00 uk-rdh-testast2 /usr/sbin/opensips[283656]: Value returned from Re-direct Script is sip:33557719673 at 10.10.4.77 2023-11-15T09:52:55.375456+00:00 uk-rdh-testast2 /usr/sbin/opensips[283656]: Target is set as sip:33557719673 at 10.10.4.77 The 302 response is below, exported from Wireshark… No. Time Source Destination Protocol Length Info 15 09:53:20.012882 10.10.12.70 10.10.14.71 SIP 666 Status: 302 Server Redirect | Frame 15: 666 bytes on wire (5328 bits), 666 bytes captured (5328 bits) Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 User Datagram Protocol, Src Port: 5070, Dst Port: 5060 Session Initiation Protocol (302) Status-Line: SIP/2.0 302 Server Redirect Message Header From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364674-1-1700041999-377836318 To: sip:+4415279xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.6023fbda08f41f7e9ebdc611f91805ef Call-ID: 55197572_103382176 at 151.x.xxx.x [Generated Call-ID: 55197572_103382176 at 151.x.xxx.x] CSeq: 61414 INVITE Via: SIP/2.0/UDP 10.10.14.71:5060;branch=z9hG4bK-6a03d713-a7b27-28f11294-7f0923fc1278 Via: SIP/2.0/UDP 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364674 Via: SIP/2.0/UDP 151.x.xxx.x:5060;branch=z9hG4bK0aB2fefabe20ff62d0e Contact: '); This pushed the closing > from the ‘Contact’ URI to the “Server” line, as you can see on the outputted 302 below… No. Time Source Destination Protocol Length Info 17 12:12:01.339034 10.10.12.70 10.10.14.71 SIP 667 Status: 302 Server Redirect | Frame 17: 667 bytes on wire (5336 bits), 667 bytes captured (5336 bits) Ethernet II, Src: bb:bb:bb:bb:bb:bb (bb:bb:bb:bb:bb:bb), Dst: aa:aa:aa:aa:aa:aa (aa:aa:aa:aa:aa:aa) Internet Protocol Version 4, Src: 10.10.12.70, Dst: 10.10.14.71 User Datagram Protocol, Src Port: 5070, Dst Port: 5060 Session Initiation Protocol (302) Status-Line: SIP/2.0 302 Server Redirect Message Header From: sip:+4478341xxxxx at 151.x.xxx.x;user=phone;tag=BN1699364667-1-1699963921-901707368 To: sip:080001xxxxx at 149.x.xxx.xx;user=phone;tag=5d8f.a1daebfe0f15d4035f73e92edd390da9 Call-ID: 55182316_87520217 at 151.x.xxx.x [Generated Call-ID: 55182316_87520217 at 151.x.xxx.x] CSeq: 706035 INVITE Via: SIP/2.0/UDP 10.10.14.71:5060;branch=z9hG4bK-34260abf-94a28-2449afc3-7f0920d0abc8 Via: SIP/2.0/UDP 149.x.xxx.xx:5060;received=151.x.xxx.x;rport=5060;branch=z9hG4bK1699364667 Via: SIP/2.0/UDP 151.x.xxx.x:5060;branch=z9hG4bK0aB220722fa7bfe603d Contact: Server: OpenSIPS (3.2.13 (x86_64/linux)) [Expert Info (Note/Undecoded): Unrecognised SIP header (>server)] [Unrecognised SIP header (>server)] [Severity level: Note] [Group: Undecoded] Content-Length: 0 Whether this is related or not, I cannot say, but thought I would include it in any case. Many Thanks, Ben From: Bogdan-Andrei Iancu Sent: Wednesday, November 15, 2023 7:30 AM To: OpenSIPS users mailling list ; Ben Bliss Subject: Re: [OpenSIPS-Users] Odd URI formatting issue when using Exec to return a variable CAUTION: This email originated from outside your organization. Exercise caution when opening attachments or clicking links, especially from unknown senders. Hi Ben, Could you post the xlog output and the resulting 302 ? Also be sure you are using the latest 3.2 version - please post the `opensips -V` here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/14/23 4:00 PM, Ben Bliss wrote: I am using Opensips v.3.2 and I am attempting to replicate the 302 redirect outlined on the blog (https://blog.opensips.org/2018/07/05/handling-sip-redirect-requests-in-realtime/), but am running into an odd issue. I am using exec to execute an external program, which returns a URI as a variable, which is then used to populate the $branch field, which is then returned as the contact URI in the 302 response. For reasons I cannot explain, even though the URI is correctly formatted in the xlogs which I print off during the call request, when the 302 message is received, the Contact URI is missing the end ‘>‘character. This means the server the 302 is being sent to doesn’t ACK the message, as the contact URI is invalid. If I specify the URI directly within OpenSIPs, then the contact header is then formatted correctly. Initially I thought the issue was related to the script returning the variable (bash script), so I swapped to a Golang program, essentially returning the same URI, but the same behaviour occurs with this as well. The script is setup as follows… if ( $si == '10.10.14.71' && is_method("INVITE") ) { exec("/usr/local/bin/sbc-route", $tU, $var(out)); xlog("Value returned from Re-direct Script is $var(out)"); $branch = $var(out); xlog("Target is set as $branch"); send_reply(302, "Server Redirect"); exit; } Can anyone offer any ideas as to why this may be occurring? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sreeram.narayanan at skit.ai Tue Nov 21 10:13:43 2023 From: sreeram.narayanan at skit.ai (Sreeram Narayanan) Date: Tue, 21 Nov 2023 15:43:43 +0530 Subject: [OpenSIPS-Users] ACK looping behind NAT In-Reply-To: <7b1fff87-d0e1-1efa-4b5f-f451d8f161a8@opensips.org> References: <7b1fff87-d0e1-1efa-4b5f-f451d8f161a8@opensips.org> Message-ID: Hi, Thanks for your response. I've added the network trace here . I've masked some of the IPs for security. This is what the ACK looks like from the OpenSIPs server. Please let me know if I need to share more information. On Wed, Nov 15, 2023 at 5:05 PM Bogdan-Andrei Iancu wrote: > Hi, > > Ideally you should provide a network capture (pcap) from the OpenSIPS > server, covering both incoming and outgoing traffic - this is the only way > to understand what is wrong with the call. > > As attachments are limited to 40K here, consider using some pastebin or > other file sharing service. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/13/23 1:26 PM, Sreeram Narayanan via Users wrote: > > Hello, > > I am trying to use OpenSIPs with the load_balancer module to balance > inbound calls between 2 Asterisk servers. The setup sits behind a NAT. The > OpenSIPs server has a public IP and a private IP. When an INVITE arrives, > it can forward it to one of the Asterisk servers and Asterisk responds with > a 200 OK. The problem starts when I receive the ACK (from Twilio). The ACK > starts bouncing between the public IP and Private IP of the OpenSIPs > server. It doesn't reach the Asterisk server and eventually times out. I > hope someone can help me with this. Thanks in advance. > > Here is my configuration: > > ####### Routing Logic ######## >> route { >> >> if (is_method("INVITE")) { >> rtpproxy_engage(); >> } >> >> if ($rm=="INVITE") { >> >> lb_start_or_next(1,"pstn"); >> } >> >> t_check_trans(); >> record_route(); >> >> t_on_failure("GW_FAILOVER"); >> >> # route the request >> if (!t_relay()) { >> sl_reply_error(); >> } >> >> exit; >> } >> >> route[RELAY] { >> if (!t_relay()) { >> sl_reply_error(); >> } >> exit; >> } >> >> failure_route[GW_FAILOVER] { >> if (t_was_cancelled()) { >> exit; >> } >> # failure detection with redirect to next available trunk >> if (t_check_status("(408)|([56][0-9][0-9])")) { >> xlog("Failed trunk $rd/$du detected \n"); >> } >> } >> > > -- > - Sreeram > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- - Sreeram -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Wed Nov 22 15:00:18 2023 From: ag at ag-projects.com (Adrian Georgescu) Date: Wed, 22 Nov 2023 12:00:18 -0300 Subject: [OpenSIPS-Users] Learning about resource lists In-Reply-To: <2779211.BddDVKsqQX@leno.mylan> References: <46085560.fMDQidcC6G@leno.mylan> <99431418-c89d-0217-132c-ee85daa35a01@opensips.org> <2779211.BddDVKsqQX@leno.mylan> Message-ID: <9DF78F48-A12A-42C3-BB7E-42F26246F600@ag-projects.com> You can try these command line scripts: Presence sip-publish-presence - PUBLISH presence to a Presence Agent sip-subscribe-winfo - SUBSCRIBE to the watcher list for given SIP address on the Presence Agent sip-subcribe-presence - SUBSCRIBE to Presence Event for a given SIP address sip-subscribe-rls - SUBSCRIBE for Presence Event to a list managed by a Resource List Server sip-subscribe-xcap-diff - SUBSCRIBE for xcap-diff Event to monitor changes to XCAP documents sip-subscribe-mwi - SUBSCRIBE for Message Waiting Indicator https://sipsimpleclient.org/testing/ Testing sipsimpleclient.org — Adrian > On 22 Nov 2023, at 11:57, Robert Dyck wrote: > > Is the RLS tutorial valid? Do we know that there are working examples? > > On Wednesday, November 22, 2023 4:01:30 A.M. PST Adrian Georgescu wrote: >> Hi Bogdan, >> >> My two cents. The reality is that adoption of XCAP is practically zero. Even >> if you build a client, you cannot make it interoperable with another, and >> XCAP was suppose to be interoperable. If I build a buddy list on one client >> and I cannot load it in another client, it makes no sense. >> >> I think that anyone building a SIP app that needs to store/fetch data on the >> SIP server can better do it using PUT/GET with a JSON, for example we took >> this path for Sylk client rather than implementing XCAP again. This is not >> interoperable between different clients, but there is no replacement >> standard for XCAP either and is much cheaper and more reliable to do it >> like this. >> >> As far as OpenSIPS is concerned one can probably make a new module or >> better, modify that existing RLS module so that it can read contacts >> directly from a database table with a schema that can be defined by the >> user. In the end what one needs is a list of URIs and a flag to see who is >> granted to see your presence, is a very simple database model. >> >> — >> Adrian >> >>> On 22 Nov 2023, at 07:02, Bogdan-Andrei Iancu wrote: >>> >>> HI Adrian, >>> >>> should we understand the everything related to xcap, like RLS, buddy list, >>> auth, etc are dropped dead at this time? if so, are you aware of any >>> replacement / alternatives here ? >>> >>> Thanks and regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/20/23 11:11 PM, Adrian Georgescu wrote: >>>> XCAP is a failure. Not that we did not try, it was a bad idea and it >>>> failed. >>>> >>>> — >>>> Adrian >>>> >>>>> On 20 Nov 2023, at 14:27, Robert Dyck > >>>>> wrote: >>>>> >>>>> The context here is subscription to presence by way of a resource list. >>>>> The learning curve is steep. I have read the tutorial. The tutorial >>>>> gives an example of a rls-service xml document. In the example the >>>>> resource list is contained within the services document. Various other >>>>> examples I have found use a separate document to hold the list. The >>>>> services document then references the list document. >>>>> >>>>> https://xcap.example.com/xcap-root/resource-lists/users/s >>>>> ip:alice at example.com /index/~~/resource-lists/list%5b at name=%22l1%22%5d>>>> esource-list> If I use an integrated server the xml documents reside in >>>>> a local database rather than the file system. Http isn't going to work. >>>>> How would one reference the database and table using rls-services >>>>> document? Or is a separate resource-lists document not supported when >>>>> using an integrated rls server? >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: logo.png Type: image/png Size: 51387 bytes Desc: not available URL: From bogdan at opensips.org Thu Nov 23 16:47:35 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Nov 2023 18:47:35 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: Message-ID: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Just checking, only the `advertised_address` global param [1], no other kind of advertising (via socket definition or script function) ? [1] https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/22/23 10:47 PM, nz deals wrote: > Hi Bogdan, > My opensips version is opensips 3.3.5 (x86_64/linux) > > Regards, > Jason > > On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu > wrote: > > What is your exact opensips version ( `opensips -V` ) ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 12:24 PM, nz deals wrote: >> Thanks Bogdan, >> >> It is due to receiving CANCEL from the caller. Yes using an >> advertised_address globally. One of our public addresses. >> >> Thanks >> >> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Jason, >> >> The CANCEL generated by OpenSIPS, is it due to a received >> CANCEL or due to an internal timeout / forking? ALso do you >> use any advertising in your setup? if yes, is it per socket, >> global or ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 7:04 AM, nz deals wrote: >>> Hi folks, >>> >>> Is there any bug in the CANCEL handling in the version 3.3.x? >>> I have a weird issue, >>> The INVITE have a VIA header as my private ip >>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>> >>> But when i CANCEL the call, the CANCEL have my public ip in >>> the VIA header. >>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>> >>> Anyway can I change the via in the cancel? >>> >>> Regards, >>> Jason >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 23 16:52:08 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Nov 2023 18:52:08 +0200 Subject: [OpenSIPS-Users] ACK looping behind NAT In-Reply-To: References: <7b1fff87-d0e1-1efa-4b5f-f451d8f161a8@opensips.org> Message-ID: Hi Sreeram, Unfortunately the ladder diagram is not enough as I cannot set the details of all the messages :(. The it looks, the 200 OK coming from 110.46.1.106:5060 may contain bogus routing information (the dialog route set), like a wrong Contact hdr point to that EXTERNAL_IP.... Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 11/21/23 12:13 PM, Sreeram Narayanan wrote: > Hi, > Thanks for your response. > > I've added the network trace here . > I've masked some of the IPs for security. > This  is what the ACK looks like > from the OpenSIPs server. Please let me know if I need to share more > information. > > On Wed, Nov 15, 2023 at 5:05 PM Bogdan-Andrei Iancu > > wrote: > > Hi, > > Ideally you should provide a network capture (pcap) from the > OpenSIPS server, covering both incoming and outgoing traffic - > this is the only way to understand what is wrong with the call. > > As attachments are limited to 40K here, consider using some > pastebin or other file sharing service. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/13/23 1:26 PM, Sreeram Narayanan via Users wrote: >> Hello, >> >> I am trying to use OpenSIPs with the load_balancer module to >> balance inbound calls between 2 Asterisk servers. The setup sits >> behind a NAT. The OpenSIPs server has a public IP and a private >> IP. When an INVITE arrives, it can forward it to one of the >> Asterisk servers and Asterisk responds with a 200 OK. The problem >> starts when I receive the ACK (from Twilio). The ACK starts >> bouncing between the public IP and Private IP of the OpenSIPs >> server. It doesn't reach the Asterisk server and eventually times >> out. I hope someone can help me with this. Thanks in advance. >> >> Here is my configuration: >> >> ####### Routing Logic ######## >> route { >> >>     if (is_method("INVITE")) { >>         rtpproxy_engage(); >>     } >> >>     if ($rm=="INVITE") { >> >>         lb_start_or_next(1,"pstn"); >>     } >> >>     t_check_trans(); >>     record_route(); >> >>     t_on_failure("GW_FAILOVER"); >> >>     # route the request >>     if (!t_relay()) { >>         sl_reply_error(); >>     } >> >>     exit; >> } >> >> route[RELAY] { >>     if (!t_relay()) { >>         sl_reply_error(); >>     } >>     exit; >> } >> >> failure_route[GW_FAILOVER] { >>     if (t_was_cancelled()) { >>         exit; >>     } >>     # failure detection with redirect to next available trunk >>     if (t_check_status("(408)|([56][0-9][0-9])")) { >>         xlog("Failed trunk $rd/$du detected \n"); >>     } >> } >> >> >> -- >> - Sreeram >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > - Sreeram -------------- next part -------------- An HTML attachment was scrubbed... URL: From rob.dyck at telus.net Thu Nov 23 19:36:10 2023 From: rob.dyck at telus.net (Robert Dyck) Date: Thu, 23 Nov 2023 11:36:10 -0800 Subject: [OpenSIPS-Users] Debug logs show To tag which are actually From tag Message-ID: <3640051.V25eIC5XRa@leno.mylan> While running opensips in debug mode I noticed that for initial requests of dialog creating methods the debug logs were showing a To tag where none actually exists. The tag displayed was actually the From tag. INVITE sip:8 at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.k7nLoQHpR;rport From: ;*tag=GSq6JzXiH * To: sip:8 at 192.168.1.2 CSeq: 21 INVITE Call-ID: VCALnmUf8V Nov 23 11:02:32 [1131200] DBG:maxfwd:is_maxfwd_present: value = 70 Nov 23 11:02:32 [1131200] *DBG:sipmsgops:has_totag: no totag * Nov 23 11:02:32 [1131200] Initial request, method is INVITE, URI is sip:8 at 192.168.1.2 Nov 23 11:02:32 [1131200] DBG:core:check_self: host != me Nov 23 11:02:32 [1131200] DBG:core:parse_headers: flags=78 Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: start searching: hash=42361, isACK=0 Nov 23 11:02:32 [1131200] DBG:tm:matching_3261: RFC3261 transaction matching failed Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: no transaction found Nov 23 11:02:32 [1131200] *DBG:core:parse_to_param: tag=GSq6JzXiH * Nov 23 11:02:32 [1131200] DBG:core:parse_to_param: end of header reached, state=11 PUBLISH sip:7 at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.myBdBZt3P;rport From: ;*tag=cbp~iz4fq* To: sip:7 at 192.168.1.2 CSeq: 21 PUBLISH Call-ID: cRC6Gw590w Max-Forwards: 70 Nov 23 11:07:08 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 Nov 23 11:07:08 [1133909] *DBG:sipmsgops:has_totag: no totag* Nov 23 11:07:08 [1133909] Initial request, method is PUBLISH, URI is sip:7 at 192.168.1.2 Nov 23 11:07:08 [1133909] DBG:core:check_self: host != me Nov 23 11:07:08 [1133909] DBG:core:parse_headers: flags=78 Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: start searching: hash=7222, isACK=0 Nov 23 11:07:08 [1133909] DBG:tm:matching_3261: RFC3261 transaction matching failed Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: no transaction found Nov 23 11:07:08 [1133909] *DBG:core:parse_to_param: tag=cbp~iz4fq* Nov 23 11:07:08 [1133909] DBG:core:parse_to_param: end of header reached, state=11 SUBSCRIBE sip:rls at 192.168.1.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.OkkDHBafu;rport From: ;*tag=g2aNdkn4o * To: sips:rls at 192.168.1.2 CSeq: 21 SUBSCRIBE Call-ID: L9LKmnZVY7 Max-Forwards: 69 Nov 23 11:07:07 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 Nov 23 11:07:07 [1133909] *DBG:sipmsgops:has_totag: no totag* Nov 23 11:07:07 [1133909] Initial request, method is SUBSCRIBE, URI is sip:rls at 192.168.1.2 Nov 23 11:07:07 [1133909] DBG:core:check_self: host != me Nov 23 11:07:07 [1133909] DBG:core:parse_headers: flags=78 Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: start searching: hash=33557, isACK=0 Nov 23 11:07:07 [1133909] DBG:tm:matching_3261: RFC3261 transaction matching failed Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: no transaction found Nov 23 11:07:07 [1133909] *DBG:core:parse_to_param: tag=g2aNdkn4o * Nov 23 11:07:07 [1133909] DBG:core:parse_to_param: end of header reached, state=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Sun Nov 26 20:28:51 2023 From: nutxase at proton.me (nutxase) Date: Sun, 26 Nov 2023 20:28:51 +0000 Subject: [OpenSIPS-Users] Websocket not responding to Bye Message-ID: <7xD5xD0XFfM6_yziR7dNEN7_exrFDOaDCQelO_OzMOZFgCSxI157aV0HJoCSFXAs0QQO5lblruHa0tF6ZxWv0haN9I7p5tslTGWrfaKXhu4=@proton.me> Hi All Having a strange issue, whereby if a call is placed between a WSS and a normal sip device or even WSS - WSS the webclient does not respond to bye [image.png] SIP to SIP calls are fine the error i see in the logs are 2023-11-24T16:40:50.024639+00:00 ip-10-254-49-96 /usr/sbin/opensips[135279]: ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5 (no corresponding listening socket) 2023-11-24T16:40:50.024804+00:00 ip-10-254-49-96 /usr/sbin/opensips[135279]: ERROR:tm:t_forward_nonack: failure to add branches any ideas what i am doing wrong? Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 14171 bytes Desc: not available URL: From nzdealshelp at gmail.com Mon Nov 27 11:44:15 2023 From: nzdealshelp at gmail.com (nz deals) Date: Tue, 28 Nov 2023 00:44:15 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: Hi Bogdan, No other advertisement via socket definition or any script function. Thank you On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu wrote: > Just checking, only the `advertised_address` global param [1], no other > kind of advertising (via socket definition or script function) ? > > [1] > https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 10:47 PM, nz deals wrote: > > Hi Bogdan, > My opensips version is opensips 3.3.5 (x86_64/linux) > > Regards, > Jason > > On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu > wrote: > >> What is your exact opensips version ( `opensips -V` ) ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 12:24 PM, nz deals wrote: >> >> Thanks Bogdan, >> >> It is due to receiving CANCEL from the caller. Yes using an >> advertised_address globally. One of our public addresses. >> >> Thanks >> >> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >> wrote: >> >>> Hi Jason, >>> >>> The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due >>> to an internal timeout / forking? ALso do you use any advertising in your >>> setup? if yes, is it per socket, global or ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 7:04 AM, nz deals wrote: >>> >>> Hi folks, >>> >>> Is there any bug in the CANCEL handling in the version 3.3.x? >>> I have a weird issue, >>> The INVITE have a VIA header as my private ip >>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>> >>> But when i CANCEL the call, the CANCEL have my public ip in the VIA >>> header. >>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>> >>> Anyway can I change the via in the cancel? >>> >>> Regards, >>> Jason >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Mon Nov 27 16:07:15 2023 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 27 Nov 2023 18:07:15 +0200 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.3, 3.3.9 and 3.2.16 Minor Releases Message-ID: <0028dc2d-9686-4fc2-827f-2469b83ddc53@opensips.org> Hi, everyone! The 3.4.3, 3.3.9 and 3.2.16 OpenSIPS minor versions are scheduled for release on *Wednesday, Dec 20th*. In preparation for the releases, starting *Wednesday, Dec 6th*, we will impose the usual *freeze* on any significant fixes (as complexity) on these stable branches, in order to ensure a /two-week safe window/ for testing. So please make sure to ping any outstanding issues on the GitHub issue tracker that may have skipped our attention.  And thank you in advance! Best regards, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 09:25:06 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 11:25:06 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: Hi, I just made a simple test using only `advertised_address` and calling from A to B and cancelling from A. The CANCEL from OpenSIPS to B obeys the advertising. I used the latest 3.3 :     version: opensips 3.3.8 (x86_64/linux) Tested with both UDP and TCP, worked in both cases. SO, try also the latest 3.3, to see if it works for you too. If not, try to put together a minimal cfg to support user registration + calls between users, cfg to show the problem. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 27.11.2023 13:44, nz deals wrote: > Hi Bogdan, > No other advertisement via socket definition or any script function. > > Thank you > > On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu > wrote: > > Just checking, only the `advertised_address` global param [1], no > other kind of advertising (via socket definition or script function) ? > > [1] > https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 11/22/23 10:47 PM, nz deals wrote: >> Hi Bogdan, >> My opensips version is opensips 3.3.5 (x86_64/linux) >> >> Regards, >> Jason >> >> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >> wrote: >> >> What is your exact opensips version ( `opensips -V` ) ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 12:24 PM, nz deals wrote: >>> Thanks Bogdan, >>> >>> It is due to receiving CANCEL from the caller. Yes using an >>> advertised_address globally. One of our public addresses. >>> >>> Thanks >>> >>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>> wrote: >>> >>> Hi Jason, >>> >>> The CANCEL generated by OpenSIPS, is it due to a >>> received CANCEL or due to an internal timeout / forking? >>> ALso do you use any advertising in your setup? if yes, >>> is it per socket, global or ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 7:04 AM, nz deals wrote: >>>> Hi folks, >>>> >>>> Is there any bug in the CANCEL handling in the version >>>> 3.3.x? >>>> I have a weird issue, >>>> The INVITE have a VIA header as my private ip >>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>> >>>> But when i CANCEL the call, the CANCEL have my public >>>> ip in the VIA header. >>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>> >>>> Anyway can I change the via in the cancel? >>>> >>>> Regards, >>>> Jason >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 09:32:53 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 11:32:53 +0200 Subject: [OpenSIPS-Users] Debug logs show To tag which are actually From tag In-Reply-To: <3640051.V25eIC5XRa@leno.mylan> References: <3640051.V25eIC5XRa@leno.mylan> Message-ID: <37fad265-bce4-48cf-975d-3c516c5fa30f@opensips.org> Hi Robert, As from SIP perspective both TO and FROM hdrs have the same syntax, internally OpenSIPS uses the same function "parse_to()" for parsing both hdrs. So, we are all good here :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 23.11.2023 21:36, Robert Dyck wrote: > > While running opensips in debug mode I noticed that for initial > requests of dialog creating methods the debug logs were showing a To > tag where none actually exists. The tag displayed was actually the > From tag. > > > INVITE sip:8 at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.k7nLoQHpR;rport > From: ;*tag=GSq6JzXiH * > To: sip:8 at 192.168.1.2 > CSeq: 21 INVITE > Call-ID: VCALnmUf8V > > > Nov 23 11:02:32 [1131200] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:02:32 [1131200] *DBG:sipmsgops:has_totag: no totag * > Nov 23 11:02:32 [1131200] Initial request, method is INVITE, URI is > sip:8 at 192.168.1.2 > > Nov 23 11:02:32 [1131200] DBG:core:check_self: host != me > Nov 23 11:02:32 [1131200] DBG:core:parse_headers: flags=78 > Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: start searching: > hash=42361, isACK=0 > Nov 23 11:02:32 [1131200] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:02:32 [1131200] *DBG:core:parse_to_param: tag=GSq6JzXiH * > Nov 23 11:02:32 [1131200] DBG:core:parse_to_param: end of header > reached, state=11 > > > PUBLISH sip:7 at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.myBdBZt3P;rport > From: ;*tag=cbp~iz4fq* > To: sip:7 at 192.168.1.2 > CSeq: 21 PUBLISH > Call-ID: cRC6Gw590w > Max-Forwards: 70 > > Nov 23 11:07:08 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:07:08 [1133909] *DBG:sipmsgops:has_totag: no totag* > Nov 23 11:07:08 [1133909] Initial request, method is PUBLISH, URI is > sip:7 at 192.168.1.2 > > Nov 23 11:07:08 [1133909] DBG:core:check_self: host != me > Nov 23 11:07:08 [1133909] DBG:core:parse_headers: flags=78 > Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: start searching: > hash=7222, isACK=0 > Nov 23 11:07:08 [1133909] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:07:08 [1133909] *DBG:core:parse_to_param: tag=cbp~iz4fq* > Nov 23 11:07:08 [1133909] DBG:core:parse_to_param: end of header > reached, state=11 > > SUBSCRIBE sip:rls at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.OkkDHBafu;rport > From: ;*tag=g2aNdkn4o * > To: sips:rls at 192.168.1.2 > CSeq: 21 SUBSCRIBE > Call-ID: L9LKmnZVY7 > Max-Forwards: 69 > > > Nov 23 11:07:07 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:07:07 [1133909] *DBG:sipmsgops:has_totag: no totag* > Nov 23 11:07:07 [1133909] Initial request, method is SUBSCRIBE, URI is > sip:rls at 192.168.1.2 > > Nov 23 11:07:07 [1133909] DBG:core:check_self: host != me > Nov 23 11:07:07 [1133909] DBG:core:parse_headers: flags=78 > Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: start searching: > hash=33557, isACK=0 > Nov 23 11:07:07 [1133909] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:07:07 [1133909] *DBG:core:parse_to_param: tag=g2aNdkn4o * > Nov 23 11:07:07 [1133909] DBG:core:parse_to_param: end of header > reached, state=1 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Tue Nov 28 09:48:38 2023 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Tue, 28 Nov 2023 22:48:38 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: Hi Bogdan, The only difference is, I am receiving traffic on UDP and sending on TCP socket. sorry i forgot to mention before the advertised_address is globally defined (public ip) Regards, Jehanzaib On Tue, Nov 28, 2023 at 10:26 PM Bogdan-Andrei Iancu wrote: > Hi, > > I just made a simple test using only `advertised_address` and calling > from A to B and cancelling from A. The CANCEL from OpenSIPS to B obeys the > advertising. I used the latest 3.3 : > version: opensips 3.3.8 (x86_64/linux) > > Tested with both UDP and TCP, worked in both cases. > > SO, try also the latest 3.3, to see if it works for you too. If not, try > to put together a minimal cfg to support user registration + calls between > users, cfg to show the problem. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 27.11.2023 13:44, nz deals wrote: > > Hi Bogdan, > No other advertisement via socket definition or any script function. > > Thank you > > On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu > wrote: > >> Just checking, only the `advertised_address` global param [1], no other >> kind of advertising (via socket definition or script function) ? >> >> [1] >> https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 10:47 PM, nz deals wrote: >> >> Hi Bogdan, >> My opensips version is opensips 3.3.5 (x86_64/linux) >> >> Regards, >> Jason >> >> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >> wrote: >> >>> What is your exact opensips version ( `opensips -V` ) ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 12:24 PM, nz deals wrote: >>> >>> Thanks Bogdan, >>> >>> It is due to receiving CANCEL from the caller. Yes using an >>> advertised_address globally. One of our public addresses. >>> >>> Thanks >>> >>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi Jason, >>>> >>>> The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due >>>> to an internal timeout / forking? ALso do you use any advertising in your >>>> setup? if yes, is it per socket, global or ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> >>>> On 11/22/23 7:04 AM, nz deals wrote: >>>> >>>> Hi folks, >>>> >>>> Is there any bug in the CANCEL handling in the version 3.3.x? >>>> I have a weird issue, >>>> The INVITE have a VIA header as my private ip >>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>> >>>> But when i CANCEL the call, the CANCEL have my public ip in the VIA >>>> header. >>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>> >>>> Anyway can I change the via in the cancel? >>>> >>>> Regards, >>>> Jason >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nzdealshelp at gmail.com Tue Nov 28 09:53:56 2023 From: nzdealshelp at gmail.com (nz deals) Date: Tue, 28 Nov 2023 22:53:56 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: Hi Bogdan, Looks like the upgrade has resolved the issue. Regards, Jason On Tue, 28 Nov 2023 at 22:25, Bogdan-Andrei Iancu wrote: > Hi, > > I just made a simple test using only `advertised_address` and calling > from A to B and cancelling from A. The CANCEL from OpenSIPS to B obeys the > advertising. I used the latest 3.3 : > version: opensips 3.3.8 (x86_64/linux) > > Tested with both UDP and TCP, worked in both cases. > > SO, try also the latest 3.3, to see if it works for you too. If not, try > to put together a minimal cfg to support user registration + calls between > users, cfg to show the problem. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 27.11.2023 13:44, nz deals wrote: > > Hi Bogdan, > No other advertisement via socket definition or any script function. > > Thank you > > On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu > wrote: > >> Just checking, only the `advertised_address` global param [1], no other >> kind of advertising (via socket definition or script function) ? >> >> [1] >> https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 10:47 PM, nz deals wrote: >> >> Hi Bogdan, >> My opensips version is opensips 3.3.5 (x86_64/linux) >> >> Regards, >> Jason >> >> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >> wrote: >> >>> What is your exact opensips version ( `opensips -V` ) ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 12:24 PM, nz deals wrote: >>> >>> Thanks Bogdan, >>> >>> It is due to receiving CANCEL from the caller. Yes using an >>> advertised_address globally. One of our public addresses. >>> >>> Thanks >>> >>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi Jason, >>>> >>>> The CANCEL generated by OpenSIPS, is it due to a received CANCEL or due >>>> to an internal timeout / forking? ALso do you use any advertising in your >>>> setup? if yes, is it per socket, global or ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> >>>> On 11/22/23 7:04 AM, nz deals wrote: >>>> >>>> Hi folks, >>>> >>>> Is there any bug in the CANCEL handling in the version 3.3.x? >>>> I have a weird issue, >>>> The INVITE have a VIA header as my private ip >>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>> >>>> But when i CANCEL the call, the CANCEL have my public ip in the VIA >>>> header. >>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>> >>>> Anyway can I change the via in the cancel? >>>> >>>> Regards, >>>> Jason >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Tue Nov 28 09:54:26 2023 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Tue, 28 Nov 2023 22:54:26 +1300 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: I have the exact same issue. I am also not able to change the Via header. Regards, Jehanzaib On Tue, Nov 28, 2023 at 10:48 PM Jehanzaib Younis wrote: > Hi Bogdan, > > The only difference is, I am receiving traffic on UDP and sending on TCP > socket. > sorry i forgot to mention before the advertised_address is > globally defined (public ip) > > > Regards, > Jehanzaib > > > On Tue, Nov 28, 2023 at 10:26 PM Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> I just made a simple test using only `advertised_address` and calling >> from A to B and cancelling from A. The CANCEL from OpenSIPS to B obeys the >> advertising. I used the latest 3.3 : >> version: opensips 3.3.8 (x86_64/linux) >> >> Tested with both UDP and TCP, worked in both cases. >> >> SO, try also the latest 3.3, to see if it works for you too. If not, try >> to put together a minimal cfg to support user registration + calls between >> users, cfg to show the problem. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 27.11.2023 13:44, nz deals wrote: >> >> Hi Bogdan, >> No other advertisement via socket definition or any script function. >> >> Thank you >> >> On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu >> wrote: >> >>> Just checking, only the `advertised_address` global param [1], no other >>> kind of advertising (via socket definition or script function) ? >>> >>> [1] >>> https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 10:47 PM, nz deals wrote: >>> >>> Hi Bogdan, >>> My opensips version is opensips 3.3.5 (x86_64/linux) >>> >>> Regards, >>> Jason >>> >>> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >>> wrote: >>> >>>> What is your exact opensips version ( `opensips -V` ) ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> >>>> On 11/22/23 12:24 PM, nz deals wrote: >>>> >>>> Thanks Bogdan, >>>> >>>> It is due to receiving CANCEL from the caller. Yes using an >>>> advertised_address globally. One of our public addresses. >>>> >>>> Thanks >>>> >>>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>>> wrote: >>>> >>>>> Hi Jason, >>>>> >>>>> The CANCEL generated by OpenSIPS, is it due to a received CANCEL or >>>>> due to an internal timeout / forking? ALso do you use any advertising in >>>>> your setup? if yes, is it per socket, global or ? >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> https://www.siphub.com >>>>> >>>>> On 11/22/23 7:04 AM, nz deals wrote: >>>>> >>>>> Hi folks, >>>>> >>>>> Is there any bug in the CANCEL handling in the version 3.3.x? >>>>> I have a weird issue, >>>>> The INVITE have a VIA header as my private ip >>>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>>> >>>>> But when i CANCEL the call, the CANCEL have my public ip in the VIA >>>>> header. >>>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>>> >>>>> Anyway can I change the via in the cancel? >>>>> >>>>> Regards, >>>>> Jason >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 10:19:03 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 12:19:03 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: I tried with UDP to TCP with the global advertising, still working. Again: 1) upgrade to latest 3.3, 2) put together a way to reproduce. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 28.11.2023 11:48, Jehanzaib Younis wrote: > Hi Bogdan, > > The only difference is, I am receiving traffic on UDP and sending on > TCP socket. > sorry i forgot to mention before the advertised_address is > globally defined (public ip) > > > Regards, > Jehanzaib > > > On Tue, Nov 28, 2023 at 10:26 PM Bogdan-Andrei Iancu > wrote: > > Hi, > > I just made a simple test using only `advertised_address` and > calling from A to B and cancelling from A. The CANCEL from > OpenSIPS to B obeys the advertising. I used the latest 3.3 : >     version: opensips 3.3.8 (x86_64/linux) > > Tested with both UDP and TCP, worked in both cases. > > SO, try also the latest 3.3, to see if it works for you too. If > not, try to put together a minimal cfg to support user > registration + calls between users, cfg to show the problem. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 27.11.2023 13:44, nz deals wrote: >> Hi Bogdan, >> No other advertisement via socket definition or any script function. >> >> Thank you >> >> On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu >> wrote: >> >> Just checking, only the `advertised_address` global param >> [1], no other kind of advertising (via socket definition or >> script function) ? >> >> [1] >> https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 10:47 PM, nz deals wrote: >>> Hi Bogdan, >>> My opensips version is opensips 3.3.5 (x86_64/linux) >>> >>> Regards, >>> Jason >>> >>> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >>> wrote: >>> >>> What is your exact opensips version ( `opensips -V` ) ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 12:24 PM, nz deals wrote: >>>> Thanks Bogdan, >>>> >>>> It is due to receiving CANCEL from the caller. Yes >>>> using an advertised_address globally. One of our public >>>> addresses. >>>> >>>> Thanks >>>> >>>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>>> wrote: >>>> >>>> Hi Jason, >>>> >>>> The CANCEL generated by OpenSIPS, is it due to a >>>> received CANCEL or due to an internal timeout / >>>> forking? ALso do you use any advertising in your >>>> setup? if yes, is it per socket, global or ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> >>>> On 11/22/23 7:04 AM, nz deals wrote: >>>>> Hi folks, >>>>> >>>>> Is there any bug in the CANCEL handling in the >>>>> version 3.3.x? >>>>> I have a weird issue, >>>>> The INVITE have a VIA header as my private ip >>>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>>> >>>>> But when i CANCEL the call, the CANCEL have my >>>>> public ip in the VIA header. >>>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>>> >>>>> Anyway can I change the via in the cancel? >>>>> >>>>> Regards, >>>>> Jason >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 10:20:35 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 12:20:35 +0200 Subject: [OpenSIPS-Users] CANCEL handling issue In-Reply-To: References: <6a74b7b1-b9eb-12ac-b7c4-bf2a77f62ceb@opensips.org> Message-ID: :+1: Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 28.11.2023 11:53, nz deals wrote: > Hi Bogdan, > > Looks like the upgrade has resolved the issue. > > Regards, > Jason > > On Tue, 28 Nov 2023 at 22:25, Bogdan-Andrei Iancu > wrote: > > Hi, > > I just made a simple test using only `advertised_address` and > calling from A to B and cancelling from A. The CANCEL from > OpenSIPS to B obeys the advertising. I used the latest 3.3 : >     version: opensips 3.3.8 (x86_64/linux) > > Tested with both UDP and TCP, worked in both cases. > > SO, try also the latest 3.3, to see if it works for you too. If > not, try to put together a minimal cfg to support user > registration + calls between users, cfg to show the problem. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 27.11.2023 13:44, nz deals wrote: >> Hi Bogdan, >> No other advertisement via socket definition or any script function. >> >> Thank you >> >> On Fri, 24 Nov 2023 at 05:47, Bogdan-Andrei Iancu >> wrote: >> >> Just checking, only the `advertised_address` global param >> [1], no other kind of advertising (via socket definition or >> script function) ? >> >> [1] >> https://www.opensips.org/Documentation/Script-CoreParameters-3-3#advertised_address >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 11/22/23 10:47 PM, nz deals wrote: >>> Hi Bogdan, >>> My opensips version is opensips 3.3.5 (x86_64/linux) >>> >>> Regards, >>> Jason >>> >>> On Thu, 23 Nov 2023 at 01:49, Bogdan-Andrei Iancu >>> wrote: >>> >>> What is your exact opensips version ( `opensips -V` ) ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> On 11/22/23 12:24 PM, nz deals wrote: >>>> Thanks Bogdan, >>>> >>>> It is due to receiving CANCEL from the caller. Yes >>>> using an advertised_address globally. One of our public >>>> addresses. >>>> >>>> Thanks >>>> >>>> On Wed, 22 Nov 2023 at 23:08, Bogdan-Andrei Iancu >>>> wrote: >>>> >>>> Hi Jason, >>>> >>>> The CANCEL generated by OpenSIPS, is it due to a >>>> received CANCEL or due to an internal timeout / >>>> forking? ALso do you use any advertising in your >>>> setup? if yes, is it per socket, global or ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> >>>> On 11/22/23 7:04 AM, nz deals wrote: >>>>> Hi folks, >>>>> >>>>> Is there any bug in the CANCEL handling in the >>>>> version 3.3.x? >>>>> I have a weird issue, >>>>> The INVITE have a VIA header as my private ip >>>>> Via: SIP/2.0/TCP 192.XX.XX.XX:5060;branch=XXXXXXX >>>>> >>>>> But when i CANCEL the call, the CANCEL have my >>>>> public ip in the VIA header. >>>>> Via: SIP/2.0/TCP 104.xx.xx.xx:5060;branch=xxxxxxx >>>>> >>>>> Anyway can I change the via in the cancel? >>>>> >>>>> Regards, >>>>> Jason >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 12:32:54 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 14:32:54 +0200 Subject: [OpenSIPS-Users] IMS OpenSIPS Working Group - Kickoff Message-ID: Hi all, As mentioned in the 3.5 planning , this release is IMS focused. And to deal with the complexity of this task, we have this new tool, the IMS OpenSIPS Working Group. And now it is the time to start its work 🙂. As starting point, we put together the Roadmap of the group and also some initial Technical Requirements (as guidance). All this info is available on the group WIKI page . The main discussion channel is the group's mailing list . So, as a fist step, if interested to contribute to the IMS work, please subscribe to the list. An now we have the official kick-off of the group by starting there the discussion on the *IMS scope in OpenSIPS*, so we need all hands on deck. Again, please subscribe to the mailing list; also check the list's archive to catch up with ongoing threads. See you on the WG-IMS list ;) -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 28 13:02:18 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Nov 2023 15:02:18 +0200 Subject: [OpenSIPS-Users] Scope of IMS in OpenSIPS - RFC Message-ID: Hi all, (disclaimer : cross lists posting is not a good practice - we will do this only to catch the attention and get momentum with this initial topic) As a first step here, is to work out the scope of the IMS implementation in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we want to understand and agree on which components of IMS may be subject of work from the OpenSIPS perspective. For example, we do consider the CSCF as a must here, but we may explore the HSS, AS, MGW or other components. From the OpenSIPS perspective, we look for IMS components which are SIP related. At least as a starting point. So, the first obvious candidate is the *Call Session Control Function (CSCF)*. And here we need to look into and address the specific functionalities of each sub-component:     * P-CSCF     * I-CSCF     * S-CSCF Again, these are the pretty obvious components, still may look into and evaluate (if of an interest of the OpenSIPS IMS implementation) areas as:     * HSS (from interconnection perspective)     * MGCF / MGW  (from interconnection perspective)     * SIP AS     * others ? Any feedback (with explanations and arguments) about what we should consider for our IMS implementation is more the welcome. I set here just a simple starting point, with no limitations or so. Feel free to contribute to the topic Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Tue Nov 28 13:08:31 2023 From: Johan at democon.be (Johan De Clercq) Date: Tue, 28 Nov 2023 14:08:31 +0100 Subject: [OpenSIPS-Users] Debug logs show To tag which are actually From tag In-Reply-To: <37fad265-bce4-48cf-975d-3c516c5fa30f@opensips.org> References: <3640051.V25eIC5XRa@leno.mylan> <37fad265-bce4-48cf-975d-3c516c5fa30f@opensips.org> Message-ID: I agree, but it's very confusing. Op di 28 nov 2023 om 10:35 schreef Bogdan-Andrei Iancu : > Hi Robert, > > As from SIP perspective both TO and FROM hdrs have the same syntax, > internally OpenSIPS uses the same function "parse_to()" for parsing both > hdrs. So, we are all good here :) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 23.11.2023 21:36, Robert Dyck wrote: > > While running opensips in debug mode I noticed that for initial requests > of dialog creating methods the debug logs were showing a To tag where none > actually exists. The tag displayed was actually the From tag. > > INVITE sip:8 at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.k7nLoQHpR;rport > From: ;*tag=GSq6JzXiH * > To: sip:8 at 192.168.1.2 > CSeq: 21 INVITE > Call-ID: VCALnmUf8V > > > Nov 23 11:02:32 [1131200] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:02:32 [1131200] *DBG:sipmsgops:has_totag: no totag * > Nov 23 11:02:32 [1131200] Initial request, method is INVITE, URI is > sip:8 at 192.168.1.2 > > Nov 23 11:02:32 [1131200] DBG:core:check_self: host != me > Nov 23 11:02:32 [1131200] DBG:core:parse_headers: flags=78 > Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: start searching: > hash=42361, isACK=0 > Nov 23 11:02:32 [1131200] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:02:32 [1131200] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:02:32 [1131200] *DBG:core:parse_to_param: tag=GSq6JzXiH * > Nov 23 11:02:32 [1131200] DBG:core:parse_to_param: end of header reached, > state=11 > > > PUBLISH sip:7 at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.myBdBZt3P;rport > From: ;*tag=cbp~iz4fq* > To: sip:7 at 192.168.1.2 > CSeq: 21 PUBLISH > Call-ID: cRC6Gw590w > Max-Forwards: 70 > > > > Nov 23 11:07:08 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:07:08 [1133909] *DBG:sipmsgops:has_totag: no totag* > Nov 23 11:07:08 [1133909] Initial request, method is PUBLISH, URI is > sip:7 at 192.168.1.2 > > > > Nov 23 11:07:08 [1133909] DBG:core:check_self: host != me > Nov 23 11:07:08 [1133909] DBG:core:parse_headers: flags=78 > Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: start searching: > hash=7222, isACK=0 > Nov 23 11:07:08 [1133909] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:07:08 [1133909] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:07:08 [1133909] *DBG:core:parse_to_param: tag=cbp~iz4fq* > Nov 23 11:07:08 [1133909] DBG:core:parse_to_param: end of header reached, > state=11 > > > > SUBSCRIBE sip:rls at 192.168.1.2 SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK.OkkDHBafu;rport > From: ;*tag=g2aNdkn4o * > To: sips:rls at 192.168.1.2 > CSeq: 21 SUBSCRIBE > Call-ID: L9LKmnZVY7 > Max-Forwards: 69 > > > Nov 23 11:07:07 [1133909] DBG:maxfwd:is_maxfwd_present: value = 70 > Nov 23 11:07:07 [1133909] *DBG:sipmsgops:has_totag: no totag* > Nov 23 11:07:07 [1133909] Initial request, method is SUBSCRIBE, URI is > sip:rls at 192.168.1.2 > > > > Nov 23 11:07:07 [1133909] DBG:core:check_self: host != me > Nov 23 11:07:07 [1133909] DBG:core:parse_headers: flags=78 > Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: start searching: > hash=33557, isACK=0 > Nov 23 11:07:07 [1133909] DBG:tm:matching_3261: RFC3261 transaction > matching failed > Nov 23 11:07:07 [1133909] DBG:tm:t_lookup_request: no transaction found > Nov 23 11:07:07 [1133909] *DBG:core:parse_to_param: tag=g2aNdkn4o * > Nov 23 11:07:07 [1133909] DBG:core:parse_to_param: end of header reached, > state=1 > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Nov 29 09:11:51 2023 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 29 Nov 2023 10:11:51 +0100 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: Message-ID: First of all: CONGRATULATIONS to the OpenSIPS community !!! (I believe this is the first step of a long and satisfying journey) On the topic: in addition to the CSCF component, I would like to see efforts on the AS (Application Server) component of the IMS infrastructure. The AS is probably way the simplest of it all, it will probably require the least modifications/additions to OpenSIPS. But I would say AS will be crucial to a lot of people/use cases. While for sure there will be a lot of cases for our community to build the voice/video complete IMS infrastructure on top of private 5G networks in enterprises and public administrations, I see as very much relevant also the use case of building infrastructure to provide additional third party services to big carriers, and to big carriers partners. Also, AS is the correct and manageable way to provide additional services even if you build the core IMS infrastructure. About HSS: this is the sancta sanctorum of a carrier/provider Apart from the venerable fraunhofer java implementation, now we can count on the flexible java implementation in https://github.com/nickvsnetworking/pyhss with a lot of features, good performances, and actually built for production. I would say better we concentrate on accessing the various different protocols of HSS (diameter/http2) from the various components (each component in IMS access HSS with a different interface with different vocabularies and actions. MGCF/MGW, if needed, will be a natural extension of our CSCF/AS architecture. Just my two cents, to keep the ball rolling, Congratulation again, -giovanni On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu wrote: > Hi all, > > (disclaimer : cross lists posting is not a good practice - we will do this > only to catch the attention and get momentum with this initial topic) > > As a first step here, is to work out the scope of the IMS implementation > in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we > want to understand and agree on which components of IMS may be subject of > work from the OpenSIPS perspective. For example, we do consider the CSCF as > a must here, but we may explore the HSS, AS, MGW or other components. > > From the OpenSIPS perspective, we look for IMS components which are SIP > related. At least as a starting point. So, the first obvious candidate is > the *Call Session Control Function (CSCF)*. And here we need to look into > and address the specific functionalities of each sub-component: > * P-CSCF > * I-CSCF > * S-CSCF > > Again, these are the pretty obvious components, still may look into and > evaluate (if of an interest of the OpenSIPS IMS implementation) areas as: > * HSS (from interconnection perspective) > * MGCF / MGW (from interconnection perspective) > * SIP AS > * others ? > > Any feedback (with explanations and arguments) about what we should > consider for our IMS implementation is more the welcome. I set here just a > simple starting point, with no limitations or so. Feel free to contribute > to the topic > > > Best regards, > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > _______________________________________________ > Wg-ims mailing list > Wg-ims at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Nov 29 09:26:07 2023 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 29 Nov 2023 10:26:07 +0100 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: Message-ID: On Wed, Nov 29, 2023 at 10:11 AM Giovanni Maruzzelli wrote: > About HSS: this is the sancta sanctorum of a carrier/provider > Apart from the venerable fraunhofer java implementation, now we can count > on the flexible java implementation in > https://github.com/nickvsnetworking/pyhss with a lot of features, good > performances, and actually built for production. > > Errata: PyHSS is obviously written in Python (not in Java at all), and very much open to integrating features/etc on GitHub -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew.colin at ipcortex.co.uk Wed Nov 29 13:15:20 2023 From: andrew.colin at ipcortex.co.uk (Andrew Colin) Date: Wed, 29 Nov 2023 13:15:20 +0000 Subject: [OpenSIPS-Users] Strange Nat issue Message-ID: Hi All, Recently deployed opensips into AWS and when we make calls between 2 webrtc clients I keep seeing this error in the logs and the call eventually drops after 32 seconds ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5 (no corresponding listening socket) ERROR:tm:t_forward_nonack: failure to add branches Normal SIP to SIP calls do not have the issue -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 29 15:17:07 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Nov 2023 17:17:07 +0200 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: References: Message-ID: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Hi Andrew, Proto 5 is WS (not WSS). Can you confirm if the error occurs in the context of the ACK ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 15:15, Andrew Colin via Users wrote: > > Hi All, > > Recently deployed opensips into AWS and when we make calls between 2 > webrtc clients I keep seeing this error in the logs and the call > eventually drops after 32 seconds > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5  (no > corresponding listening socket) > > ERROR:tm:t_forward_nonack: failure to add branches > > Normal SIP to SIP calls do not have the issue > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew.colin at ipcortex.co.uk Wed Nov 29 15:27:31 2023 From: andrew.colin at ipcortex.co.uk (Andrew Colin) Date: Wed, 29 Nov 2023 15:27:31 +0000 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> References: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Message-ID: Hi Bogdan, Seems to be in the context of the ACK yes. Why would I be seeing proto 5 if we are using WSS then? Kind Regards From: Bogdan-Andrei Iancu Date: Wednesday, 29 November 2023 at 15:17 To: users at lists.opensips.org , Andrew Colin Subject: Re: [OpenSIPS-Users] Strange Nat issue Hi Andrew, Proto 5 is WS (not WSS). Can you confirm if the error occurs in the context of the ACK ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 15:15, Andrew Colin via Users wrote: Hi All, Recently deployed opensips into AWS and when we make calls between 2 webrtc clients I keep seeing this error in the logs and the call eventually drops after 32 seconds ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5 (no corresponding listening socket) ERROR:tm:t_forward_nonack: failure to add branches Normal SIP to SIP calls do not have the issue _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 29 15:33:28 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Nov 2023 17:33:28 +0200 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: References: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Message-ID: The routing of the ACK is done accordingly to the routing info in the ACK itself (like RURI and Route hdrs). To see which is the next hop (as SIP for the ACK), after the successful loose_route(), log the $ru and $du... And I understand you are actually using WSS, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 17:27, Andrew Colin wrote: > > Hi Bogdan, > > Seems to be in the context of the ACK yes. > > Why would I be seeing proto 5 if we are using WSS then? > > Kind Regards > > *From: *Bogdan-Andrei Iancu > *Date: *Wednesday, 29 November 2023 at 15:17 > *To: *users at lists.opensips.org , Andrew > Colin > *Subject: *Re: [OpenSIPS-Users] Strange Nat issue > > Hi Andrew, > > Proto 5 is WS (not WSS). Can you confirm if the error occurs in the > context of the ACK ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 15:15, Andrew Colin via Users wrote: > > Hi All, > > Recently deployed opensips into AWS and when we make calls between > 2 webrtc clients I keep seeing this error in the logs and the call > eventually drops after 32 seconds > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5  (no > corresponding listening socket) > > ERROR:tm:t_forward_nonack: failure to add branches > > Normal SIP to SIP calls do not have the issue > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew.colin at ipcortex.co.uk Wed Nov 29 15:37:57 2023 From: andrew.colin at ipcortex.co.uk (Andrew Colin) Date: Wed, 29 Nov 2023 15:37:57 +0000 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: References: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Message-ID: Correct I am using WSS I have tested with SIP as well and had no issues From: Bogdan-Andrei Iancu Date: Wednesday, 29 November 2023 at 15:33 To: Andrew Colin , users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Strange Nat issue The routing of the ACK is done accordingly to the routing info in the ACK itself (like RURI and Route hdrs). To see which is the next hop (as SIP for the ACK), after the successful loose_route(), log the $ru and $du... And I understand you are actually using WSS, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 17:27, Andrew Colin wrote: Hi Bogdan, Seems to be in the context of the ACK yes. Why would I be seeing proto 5 if we are using WSS then? Kind Regards From: Bogdan-Andrei Iancu Date: Wednesday, 29 November 2023 at 15:17 To: users at lists.opensips.org , Andrew Colin Subject: Re: [OpenSIPS-Users] Strange Nat issue Hi Andrew, Proto 5 is WS (not WSS). Can you confirm if the error occurs in the context of the ACK ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 15:15, Andrew Colin via Users wrote: Hi All, Recently deployed opensips into AWS and when we make calls between 2 webrtc clients I keep seeing this error in the logs and the call eventually drops after 32 seconds ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5 (no corresponding listening socket) ERROR:tm:t_forward_nonack: failure to add branches Normal SIP to SIP calls do not have the issue _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 29 15:38:13 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Nov 2023 17:38:13 +0200 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: Message-ID: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Hi Giovanni, Thanks for the feedback here, a valuable one as usual :). On the HSS, what you are saying aligns with the my own thoughts - that its functioning logic is somehow outside the our scope here, but we need to pay attention to the interfacing (DIAMETER or HTTP2.0). Now, on the AS side - as I understand, it holds whatever custom logic the operator may have in routing and proving services (included VAS's). So to say, I see it as a highly programmable component. And if so, what we need to provide here is probably a very high level interface / API to allow call manipulation in a very abstract way... :-/ ?? Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 11:11, Giovanni Maruzzelli wrote: > First of all: > CONGRATULATIONS to the OpenSIPS community !!! > (I believe this is the first step of a long and satisfying journey) > > On the topic: > in addition to the CSCF component, I would like to see efforts on the > AS (Application Server) component of the IMS infrastructure. > > The AS is probably way the simplest of it all, it will probably > require the least modifications/additions to OpenSIPS. > > But I would say AS will be crucial to a lot of people/use cases. > > While for sure there will be a lot of cases for our community to build > the voice/video complete IMS infrastructure on top of private 5G > networks in enterprises and public administrations, I see as very much > relevant also the use case of building infrastructure to provide > additional third party services to big carriers, and to big carriers > partners. > > Also, AS is the correct and manageable way to provide additional > services even if you build the core IMS infrastructure. > > About HSS: this is the sancta sanctorum of a carrier/provider > Apart from the venerable fraunhofer java implementation, now we can > count on the flexible java implementation in > https://github.com/nickvsnetworking/pyhss with a lot of features, good > performances, and actually built for production. > > I would say better we concentrate on accessing the various different > protocols of HSS (diameter/http2) from the various components (each > component in IMS access HSS with a different interface with > different vocabularies and actions. > > MGCF/MGW, if needed, will be a natural extension of our CSCF/AS > architecture. > > Just my two cents, to keep the ball rolling, > > Congratulation again, > > -giovanni > > > On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu > wrote: > > Hi all, > > (disclaimer : cross lists posting is not a good practice - we will > do this only to catch the attention and get momentum with this > initial topic) > > As a first step here, is to work out the scope of the IMS > implementation in OpenSIPS. IMS is a vast concept, with SIP and > non-SIP components, and we want to understand and agree on which > components of IMS may be subject of work from the OpenSIPS > perspective. For example, we do consider the CSCF as a must here, > but we may explore the HSS, AS, MGW or other components. > > From the OpenSIPS perspective, we look for IMS components which > are SIP related. At least as a starting point. So, the first > obvious candidate is the *Call Session Control Function (CSCF)*. > And here we need to look into and address the specific > functionalities of each sub-component: >     * P-CSCF >     * I-CSCF >     * S-CSCF > > Again, these are the pretty obvious components, still may look > into and evaluate (if of an interest of the OpenSIPS IMS > implementation) areas as: >     * HSS (from interconnection perspective) >     * MGCF / MGW  (from interconnection perspective) >     * SIP AS >     * others ? > > Any feedback (with explanations and arguments) about what we > should consider for our IMS implementation is more the welcome. I > set here just a simple starting point, with no limitations or so. > Feel free to contribute to the topic > > > Best regards, > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > _______________________________________________ > Wg-ims mailing list > Wg-ims at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Nov 29 16:07:36 2023 From: Johan at democon.be (Johan De Clercq) Date: Wed, 29 Nov 2023 17:07:36 +0100 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> References: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Message-ID: In addition, the IMS should be able to handle 4G and 5G calls. In my opinion, we should no longer about 2 and 3 G as they are being phased out everywhere. wkr, Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu : > Hi Giovanni, > > Thanks for the feedback here, a valuable one as usual :). > > On the HSS, what you are saying aligns with the my own thoughts - that its > functioning logic is somehow outside the our scope here, but we need to pay > attention to the interfacing (DIAMETER or HTTP2.0). > > Now, on the AS side - as I understand, it holds whatever custom logic the > operator may have in routing and proving services (included VAS's). So to > say, I see it as a highly programmable component. And if so, what we need > to provide here is probably a very high level interface / API to allow call > manipulation in a very abstract way... :-/ ?? > > Best Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 11:11, Giovanni Maruzzelli wrote: > > First of all: > CONGRATULATIONS to the OpenSIPS community !!! > (I believe this is the first step of a long and satisfying journey) > > On the topic: > in addition to the CSCF component, I would like to see efforts on the AS > (Application Server) component of the IMS infrastructure. > > The AS is probably way the simplest of it all, it will probably require > the least modifications/additions to OpenSIPS. > > But I would say AS will be crucial to a lot of people/use cases. > > While for sure there will be a lot of cases for our community to build the > voice/video complete IMS infrastructure on top of private 5G networks in > enterprises and public administrations, I see as very much relevant also > the use case of building infrastructure to provide additional third party > services to big carriers, and to big carriers partners. > > Also, AS is the correct and manageable way to provide additional services > even if you build the core IMS infrastructure. > > About HSS: this is the sancta sanctorum of a carrier/provider > Apart from the venerable fraunhofer java implementation, now we can count > on the flexible java implementation in > https://github.com/nickvsnetworking/pyhss with a lot of features, good > performances, and actually built for production. > > I would say better we concentrate on accessing the various different > protocols of HSS (diameter/http2) from the various components (each > component in IMS access HSS with a different interface with > different vocabularies and actions. > > MGCF/MGW, if needed, will be a natural extension of our CSCF/AS > architecture. > > Just my two cents, to keep the ball rolling, > > Congratulation again, > > -giovanni > > > On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu > wrote: > >> Hi all, >> >> (disclaimer : cross lists posting is not a good practice - we will do >> this only to catch the attention and get momentum with this initial topic) >> >> As a first step here, is to work out the scope of the IMS implementation >> in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we >> want to understand and agree on which components of IMS may be subject of >> work from the OpenSIPS perspective. For example, we do consider the CSCF as >> a must here, but we may explore the HSS, AS, MGW or other components. >> >> From the OpenSIPS perspective, we look for IMS components which are SIP >> related. At least as a starting point. So, the first obvious candidate is >> the *Call Session Control Function (CSCF)*. And here we need to look >> into and address the specific functionalities of each sub-component: >> * P-CSCF >> * I-CSCF >> * S-CSCF >> >> Again, these are the pretty obvious components, still may look into and >> evaluate (if of an interest of the OpenSIPS IMS implementation) areas as: >> * HSS (from interconnection perspective) >> * MGCF / MGW (from interconnection perspective) >> * SIP AS >> * others ? >> >> Any feedback (with explanations and arguments) about what we should >> consider for our IMS implementation is more the welcome. I set here just a >> simple starting point, with no limitations or so. Feel free to contribute >> to the topic >> >> >> Best regards, >> >> -- >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> _______________________________________________ >> Wg-ims mailing list >> Wg-ims at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >> > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Wg-ims mailing list > Wg-ims at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 29 17:23:05 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Nov 2023 19:23:05 +0200 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: References: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Message-ID: <65b0d1d4-dc75-4917-b989-39cd829d3f0b@opensips.org> Have you tried the xlog'ing I suggested? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 18:46, Andrew Colin wrote: > > Could this be an issue with my websocket connection? > > *From: *Bogdan-Andrei Iancu > *Date: *Wednesday, 29 November 2023 at 15:33 > *To: *Andrew Colin , > users at lists.opensips.org > *Subject: *Re: [OpenSIPS-Users] Strange Nat issue > > The routing of the ACK is done accordingly to the routing info in the > ACK itself (like RURI and Route hdrs). To see which is the next hop > (as SIP for the ACK), after the successful loose_route(), log the $ru > and $du... And I understand you are actually using WSS, right ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 17:27, Andrew Colin wrote: > > Hi Bogdan, > > Seems to be in the context of the ACK yes. > > Why would I be seeing proto 5 if we are using WSS then? > > Kind Regards > > *From: *Bogdan-Andrei Iancu > > *Date: *Wednesday, 29 November 2023 at 15:17 > *To: *users at lists.opensips.org > , Andrew Colin > > *Subject: *Re: [OpenSIPS-Users] Strange Nat issue > > Hi Andrew, > > Proto 5 is WS (not WSS). Can you confirm if the error occurs in > the context of the ACK ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 29.11.2023 15:15, Andrew Colin via Users wrote: > > Hi All, > > Recently deployed opensips into AWS and when we make calls > between 2 webrtc clients I keep seeing this error in the logs > and the call eventually drops after 32 seconds > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5  (no > corresponding listening socket) > > ERROR:tm:t_forward_nonack: failure to add branches > > Normal SIP to SIP calls do not have the issue > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Nov 29 17:25:25 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Nov 2023 19:25:25 +0200 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Message-ID: Hi Johan, The lowest point we should address in the whole IMS arch is the P-CSCF, so we are agnostic to the actual transport layer below us (like the xG stuff). Or am I saying here something wrong and there are some implications to the upper layers ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 18:07, Johan De Clercq wrote: > In addition, the IMS should be able to handle 4G and 5G calls. > In my opinion, we should no longer about 2 and 3 G as they are being > phased out everywhere. > > wkr, > > Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu > : > > Hi Giovanni, > > Thanks for the feedback here, a valuable one as usual :). > > On the HSS, what you are saying aligns with the my own thoughts - > that its functioning logic is somehow outside the our scope here, > but we need to pay attention to the interfacing (DIAMETER or HTTP2.0). > > Now, on the AS side - as I understand, it holds whatever custom > logic the operator may have in routing and proving services > (included VAS's). So to say, I see it as a highly programmable > component. And if so, what we need to provide here is probably a > very high level interface / API to allow call manipulation in a > very abstract way... :-/ ?? > > Best Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 11:11, Giovanni Maruzzelli wrote: >> First of all: >> CONGRATULATIONS to the OpenSIPS community !!! >> (I believe this is the first step of a long and satisfying journey) >> >> On the topic: >> in addition to the CSCF component, I would like to see efforts on >> the AS (Application Server) component of the IMS infrastructure. >> >> The AS is probably way the simplest of it all, it will probably >> require the least modifications/additions to OpenSIPS. >> >> But I would say AS will be crucial to a lot of people/use cases. >> >> While for sure there will be a lot of cases for our community to >> build the voice/video complete IMS infrastructure on top of >> private 5G networks in enterprises and public administrations, I >> see as very much relevant also the use case of building >> infrastructure to provide additional third party services to big >> carriers, and to big carriers partners. >> >> Also, AS is the correct and manageable way to provide additional >> services even if you build the core IMS infrastructure. >> >> About HSS: this is the sancta sanctorum of a carrier/provider >> Apart from the venerable fraunhofer java implementation, now we >> can count on the flexible java implementation in >> https://github.com/nickvsnetworking/pyhss with a lot of features, >> good performances, and actually built for production. >> >> I would say better we concentrate on accessing the various >> different protocols of HSS (diameter/http2) from the various >> components (each component in IMS access HSS with a different >> interface with different vocabularies and actions. >> >> MGCF/MGW, if needed, will be a natural extension of our CSCF/AS >> architecture. >> >> Just my two cents, to keep the ball rolling, >> >> Congratulation again, >> >> -giovanni >> >> >> On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu >> wrote: >> >> Hi all, >> >> (disclaimer : cross lists posting is not a good practice - we >> will do this only to catch the attention and get momentum >> with this initial topic) >> >> As a first step here, is to work out the scope of the IMS >> implementation in OpenSIPS. IMS is a vast concept, with SIP >> and non-SIP components, and we want to understand and agree >> on which components of IMS may be subject of work from the >> OpenSIPS perspective. For example, we do consider the CSCF as >> a must here, but we may explore the HSS, AS, MGW or other >> components. >> >> From the OpenSIPS perspective, we look for IMS components >> which are SIP related. At least as a starting point. So, the >> first obvious candidate is the *Call Session Control Function >> (CSCF)*. And here we need to look into and address the >> specific functionalities of each sub-component: >>     * P-CSCF >>     * I-CSCF >>     * S-CSCF >> >> Again, these are the pretty obvious components, still may >> look into and evaluate (if of an interest of the OpenSIPS IMS >> implementation) areas as: >>     * HSS (from interconnection perspective) >>     * MGCF / MGW (from interconnection perspective) >>     * SIP AS >>     * others ? >> >> Any feedback (with explanations and arguments) about what we >> should consider for our IMS implementation is more the >> welcome. I set here just a simple starting point, with no >> limitations or so. Feel free to contribute to the topic >> >> >> Best regards, >> >> -- >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> _______________________________________________ >> Wg-ims mailing list >> Wg-ims at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> > > _______________________________________________ > Wg-ims mailing list > Wg-ims at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Nov 29 17:45:58 2023 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 29 Nov 2023 18:45:58 +0100 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Message-ID: Yes, actually there is a difference between 5g and 4g infrastructure, that actually often involve different interfacing from IMS to it, particularly pcscf and icscf, eg: the way they interact with hss and pcf/pcrf. Problem is that 4g infrastructure is different from 5g. When they implement 4g+5g, they implement actually both (so, no problem) 4g+5g is called NSA (not stand alone) A pure 5g is SA (stand alone) and offer different interfaces from the ones provided by 4g. In NSA you (IMS) can behave like it's pure 4g (you use 4g interfaces to do all things, even for the 5g part) In SA not at all, you must interface to 5g The main difference for what ims is concerned is pcf vs pcrf Let's note that most private networks (enterprise, etc) will be SA Most carriers will obviously be NSA answered from mobile, please pardon terseness and typos, -giovanni On Wed, Nov 29, 2023, 18:25 Bogdan-Andrei Iancu wrote: > Hi Johan, > > The lowest point we should address in the whole IMS arch is the P-CSCF, so > we are agnostic to the actual transport layer below us (like the xG stuff). > Or am I saying here something wrong and there are some implications to the > upper layers ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 18:07, Johan De Clercq wrote: > > In addition, the IMS should be able to handle 4G and 5G calls. > In my opinion, we should no longer about 2 and 3 G as they are being > phased out everywhere. > > wkr, > > Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu < > bogdan at opensips.org>: > >> Hi Giovanni, >> >> Thanks for the feedback here, a valuable one as usual :). >> >> On the HSS, what you are saying aligns with the my own thoughts - that >> its functioning logic is somehow outside the our scope here, but we need to >> pay attention to the interfacing (DIAMETER or HTTP2.0). >> >> Now, on the AS side - as I understand, it holds whatever custom logic the >> operator may have in routing and proving services (included VAS's). So to >> say, I see it as a highly programmable component. And if so, what we need >> to provide here is probably a very high level interface / API to allow call >> manipulation in a very abstract way... :-/ ?? >> >> Best Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 29.11.2023 11:11, Giovanni Maruzzelli wrote: >> >> First of all: >> CONGRATULATIONS to the OpenSIPS community !!! >> (I believe this is the first step of a long and satisfying journey) >> >> On the topic: >> in addition to the CSCF component, I would like to see efforts on the AS >> (Application Server) component of the IMS infrastructure. >> >> The AS is probably way the simplest of it all, it will probably require >> the least modifications/additions to OpenSIPS. >> >> But I would say AS will be crucial to a lot of people/use cases. >> >> While for sure there will be a lot of cases for our community to build >> the voice/video complete IMS infrastructure on top of private 5G networks >> in enterprises and public administrations, I see as very much relevant also >> the use case of building infrastructure to provide additional third party >> services to big carriers, and to big carriers partners. >> >> Also, AS is the correct and manageable way to provide additional services >> even if you build the core IMS infrastructure. >> >> About HSS: this is the sancta sanctorum of a carrier/provider >> Apart from the venerable fraunhofer java implementation, now we can count >> on the flexible java implementation in >> https://github.com/nickvsnetworking/pyhss with a lot of features, good >> performances, and actually built for production. >> >> I would say better we concentrate on accessing the various different >> protocols of HSS (diameter/http2) from the various components (each >> component in IMS access HSS with a different interface with >> different vocabularies and actions. >> >> MGCF/MGW, if needed, will be a natural extension of our CSCF/AS >> architecture. >> >> Just my two cents, to keep the ball rolling, >> >> Congratulation again, >> >> -giovanni >> >> >> On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu >> wrote: >> >>> Hi all, >>> >>> (disclaimer : cross lists posting is not a good practice - we will do >>> this only to catch the attention and get momentum with this initial topic) >>> >>> As a first step here, is to work out the scope of the IMS implementation >>> in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we >>> want to understand and agree on which components of IMS may be subject of >>> work from the OpenSIPS perspective. For example, we do consider the CSCF as >>> a must here, but we may explore the HSS, AS, MGW or other components. >>> >>> From the OpenSIPS perspective, we look for IMS components which are SIP >>> related. At least as a starting point. So, the first obvious candidate is >>> the *Call Session Control Function (CSCF)*. And here we need to look >>> into and address the specific functionalities of each sub-component: >>> * P-CSCF >>> * I-CSCF >>> * S-CSCF >>> >>> Again, these are the pretty obvious components, still may look into and >>> evaluate (if of an interest of the OpenSIPS IMS implementation) areas as: >>> * HSS (from interconnection perspective) >>> * MGCF / MGW (from interconnection perspective) >>> * SIP AS >>> * others ? >>> >>> Any feedback (with explanations and arguments) about what we should >>> consider for our IMS implementation is more the welcome. I set here just a >>> simple starting point, with no limitations or so. Feel free to contribute >>> to the topic >>> >>> >>> Best regards, >>> >>> -- >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> _______________________________________________ >>> Wg-ims mailing list >>> Wg-ims at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >>> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> >> _______________________________________________ >> Wg-ims mailing list >> Wg-ims at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Wed Nov 29 17:52:48 2023 From: johan at democon.be (Johan De Clercq) Date: Wed, 29 Nov 2023 17:52:48 +0000 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Message-ID: I agree Giovanni. If the decision is made to go nsa (even not in first release), that should be taken into account. That’s why we need to scope. Verzonden vanuit Outlook voor iOS ________________________________ Van: Giovanni Maruzzelli Verzonden: Wednesday, November 29, 2023 6:45:58 PM Aan: Bogdan-Andrei Iancu CC: Johan De Clercq ; Giovanni Maruzzelli ; wg-ims at lists.opensips.org ; OpenSIPS users mailling list Onderwerp: Re: [WG-IMS] Scope of IMS in OpenSIPS - RFC Yes, actually there is a difference between 5g and 4g infrastructure, that actually often involve different interfacing from IMS to it, particularly pcscf and icscf, eg: the way they interact with hss and pcf/pcrf. Problem is that 4g infrastructure is different from 5g. When they implement 4g+5g, they implement actually both (so, no problem) 4g+5g is called NSA (not stand alone) A pure 5g is SA (stand alone) and offer different interfaces from the ones provided by 4g. In NSA you (IMS) can behave like it's pure 4g (you use 4g interfaces to do all things, even for the 5g part) In SA not at all, you must interface to 5g The main difference for what ims is concerned is pcf vs pcrf Let's note that most private networks (enterprise, etc) will be SA Most carriers will obviously be NSA answered from mobile, please pardon terseness and typos, -giovanni On Wed, Nov 29, 2023, 18:25 Bogdan-Andrei Iancu > wrote: Hi Johan, The lowest point we should address in the whole IMS arch is the P-CSCF, so we are agnostic to the actual transport layer below us (like the xG stuff). Or am I saying here something wrong and there are some implications to the upper layers ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 18:07, Johan De Clercq wrote: In addition, the IMS should be able to handle 4G and 5G calls. In my opinion, we should no longer about 2 and 3 G as they are being phased out everywhere. wkr, Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu >: Hi Giovanni, Thanks for the feedback here, a valuable one as usual :). On the HSS, what you are saying aligns with the my own thoughts - that its functioning logic is somehow outside the our scope here, but we need to pay attention to the interfacing (DIAMETER or HTTP2.0). Now, on the AS side - as I understand, it holds whatever custom logic the operator may have in routing and proving services (included VAS's). So to say, I see it as a highly programmable component. And if so, what we need to provide here is probably a very high level interface / API to allow call manipulation in a very abstract way... :-/ ?? Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 11:11, Giovanni Maruzzelli wrote: First of all: CONGRATULATIONS to the OpenSIPS community !!! (I believe this is the first step of a long and satisfying journey) On the topic: in addition to the CSCF component, I would like to see efforts on the AS (Application Server) component of the IMS infrastructure. The AS is probably way the simplest of it all, it will probably require the least modifications/additions to OpenSIPS. But I would say AS will be crucial to a lot of people/use cases. While for sure there will be a lot of cases for our community to build the voice/video complete IMS infrastructure on top of private 5G networks in enterprises and public administrations, I see as very much relevant also the use case of building infrastructure to provide additional third party services to big carriers, and to big carriers partners. Also, AS is the correct and manageable way to provide additional services even if you build the core IMS infrastructure. About HSS: this is the sancta sanctorum of a carrier/provider Apart from the venerable fraunhofer java implementation, now we can count on the flexible java implementation in https://github.com/nickvsnetworking/pyhss with a lot of features, good performances, and actually built for production. I would say better we concentrate on accessing the various different protocols of HSS (diameter/http2) from the various components (each component in IMS access HSS with a different interface with different vocabularies and actions. MGCF/MGW, if needed, will be a natural extension of our CSCF/AS architecture. Just my two cents, to keep the ball rolling, Congratulation again, -giovanni On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu > wrote: Hi all, (disclaimer : cross lists posting is not a good practice - we will do this only to catch the attention and get momentum with this initial topic) As a first step here, is to work out the scope of the IMS implementation in OpenSIPS. IMS is a vast concept, with SIP and non-SIP components, and we want to understand and agree on which components of IMS may be subject of work from the OpenSIPS perspective. For example, we do consider the CSCF as a must here, but we may explore the HSS, AS, MGW or other components. From the OpenSIPS perspective, we look for IMS components which are SIP related. At least as a starting point. So, the first obvious candidate is the Call Session Control Function (CSCF). And here we need to look into and address the specific functionalities of each sub-component: * P-CSCF * I-CSCF * S-CSCF Again, these are the pretty obvious components, still may look into and evaluate (if of an interest of the OpenSIPS IMS implementation) areas as: * HSS (from interconnection perspective) * MGCF / MGW (from interconnection perspective) * SIP AS * others ? Any feedback (with explanations and arguments) about what we should consider for our IMS implementation is more the welcome. I set here just a simple starting point, with no limitations or so. Feel free to contribute to the topic Best regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com _______________________________________________ Wg-ims mailing list Wg-ims at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 _______________________________________________ Wg-ims mailing list Wg-ims at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 29 19:14:13 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Wed, 29 Nov 2023 21:14:13 +0200 Subject: [OpenSIPS-Users] Strange Nat issue In-Reply-To: References: <2d3c557f-6b8b-4b2c-a955-4a711812fdd7@opensips.org> Message-ID: Hi, Andrew! What WebRTC client are you using? Could you capture the SIP messages exchanged between the two endpoints? Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 11/29/23 17:37, Andrew Colin via Users wrote: > Correct I am using WSS > > I have tested with SIP as well and had no issues > > *From: *Bogdan-Andrei Iancu > *Date: *Wednesday, 29 November 2023 at 15:33 > *To: *Andrew Colin , > users at lists.opensips.org > *Subject: *Re: [OpenSIPS-Users] Strange Nat issue > > The routing of the ACK is done accordingly to the routing info in the > ACK itself (like RURI and Route hdrs). To see which is the next hop (as > SIP for the ACK), after the successful loose_route(), log the $ru and > $du... And I understand you are actually using WSS, right ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 29.11.2023 17:27, Andrew Colin wrote: > > Hi Bogdan, > > Seems to be in the context of the ACK yes. > > Why would I be seeing proto 5 if we are using WSS then? > > Kind Regards > > *From: *Bogdan-Andrei Iancu > > *Date: *Wednesday, 29 November 2023 at 15:17 > *To: *users at lists.opensips.org > , Andrew > Colin > *Subject: *Re: [OpenSIPS-Users] Strange Nat issue > > Hi Andrew, > > Proto 5 is WS (not WSS). Can you confirm if the error occurs in the > context of the ACK ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > https://www.opensips-solutions.com > > https://www.siphub.com > > On 29.11.2023 15:15, Andrew Colin via Users wrote: > > Hi All, > > Recently deployed opensips into AWS and when we make calls > between 2 webrtc clients I keep seeing this error in the logs > and the call eventually drops after 32 seconds > > ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 5  (no > corresponding listening socket) > > ERROR:tm:t_forward_nonack: failure to add branches > > Normal SIP to SIP calls do not have the issue > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From greg at switchtel.co.za Thu Nov 30 00:34:59 2023 From: greg at switchtel.co.za (Gregory Massel) Date: Thu, 30 Nov 2023 02:34:59 +0200 Subject: [OpenSIPS-Users] Help dropping SQL injection attacks Message-ID: Hi all I'm wondering what the best practice is in terms of detection and dropping attempted SQL injection attacks? Is something like the following adequate or can this be enhanced: if ( $fU != $(fU{s.escape.common}) || $tU != $(tU{s.escape.common}) ) { drop(); } Obviously this does not remove the need to escape anything passed to avp_db_query(), however, what I want to do is identify these sorts of attacks at the top of the script and avoid processing. To date all the attacks I've seen focus on the contact and from user, e.g.: INVITEsip:00111390237920793 at x.x.x.x:5060;transport=UDP SIP/2.0 Contact: To: From:;tag=v2pjtxqb I'm not quite sure how to match the Contact user. Would the following work? if ( $(ct.fields(uri){uri.user}) != $(ct.fields(uri){uri.user}{s.escape.common}) ) { drop(); } -- Regards *Gregory Massel* -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Nov 30 09:01:25 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Nov 2023 11:01:25 +0200 Subject: [OpenSIPS-Users] [WG-IMS] Scope of IMS in OpenSIPS - RFC In-Reply-To: References: <0d7d8cd9-8070-43af-babf-d5a83dc002cb@opensips.org> Message-ID: It looks I have to do more reading in this direction...some homework :). But yes, it seems a good point to be taken into consideration. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 29.11.2023 19:45, Giovanni Maruzzelli wrote: > Yes, actually there is a difference between 5g and 4g infrastructure, > that actually often involve different interfacing from IMS to it, > particularly pcscf and icscf, eg: the way they interact with hss and > pcf/pcrf. > Problem is that 4g infrastructure is different from 5g. When they > implement 4g+5g, they implement actually both (so, no problem) > > 4g+5g is called NSA (not stand alone) > > A pure 5g is SA (stand alone) and offer different interfaces from the > ones provided by 4g. > > In NSA you (IMS) can behave like it's pure 4g (you use 4g interfaces > to do all things, even for the 5g part) > > In SA not at all, you must interface to 5g > > The main difference for what ims is concerned is pcf vs pcrf > > Let's note that most private networks (enterprise, etc) will be SA > > Most carriers will obviously be NSA > > > > answered from mobile, please pardon terseness and typos, > -giovanni > > On Wed, Nov 29, 2023, 18:25 Bogdan-Andrei Iancu > wrote: > > Hi Johan, > > The lowest point we should address in the whole IMS arch is the > P-CSCF, so we are agnostic to the actual transport layer below us > (like the xG stuff). Or am I saying here something wrong and there > are some implications to the upper layers ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 29.11.2023 18:07, Johan De Clercq wrote: >> In addition, the IMS should be able to handle 4G and 5G calls. >> In my opinion, we should no longer about 2 and 3 G as they are >> being phased out everywhere. >> >> wkr, >> >> Op wo 29 nov 2023 om 16:39 schreef Bogdan-Andrei Iancu >> : >> >> Hi Giovanni, >> >> Thanks for the feedback here, a valuable one as usual :). >> >> On the HSS, what you are saying aligns with the my own >> thoughts - that its functioning logic is somehow outside the >> our scope here, but we need to pay attention to the >> interfacing (DIAMETER or HTTP2.0). >> >> Now, on the AS side - as I understand, it holds whatever >> custom logic the operator may have in routing and proving >> services (included VAS's). So to say, I see it as a highly >> programmable component. And if so, what we need to provide >> here is probably a very high level interface / API to allow >> call manipulation in a very abstract way... :-/ ?? >> >> Best Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 29.11.2023 11:11, Giovanni Maruzzelli wrote: >>> First of all: >>> CONGRATULATIONS to the OpenSIPS community !!! >>> (I believe this is the first step of a long and >>> satisfying journey) >>> >>> On the topic: >>> in addition to the CSCF component, I would like to see >>> efforts on the AS (Application Server) component of the IMS >>> infrastructure. >>> >>> The AS is probably way the simplest of it all, it will >>> probably require the least modifications/additions to OpenSIPS. >>> >>> But I would say AS will be crucial to a lot of people/use cases. >>> >>> While for sure there will be a lot of cases for our >>> community to build the voice/video complete IMS >>> infrastructure on top of private 5G networks in enterprises >>> and public administrations, I see as very much relevant also >>> the use case of building infrastructure to provide >>> additional third party services to big carriers, and to big >>> carriers partners. >>> >>> Also, AS is the correct and manageable way to provide >>> additional services even if you build the core IMS >>> infrastructure. >>> >>> About HSS: this is the sancta sanctorum of a carrier/provider >>> Apart from the venerable fraunhofer java implementation, now >>> we can count on the flexible java implementation in >>> https://github.com/nickvsnetworking/pyhss with a lot of >>> features, good performances, and actually built for production. >>> >>> I would say better we concentrate on accessing the various >>> different protocols of HSS (diameter/http2) from the various >>> components (each component in IMS access HSS with a >>> different interface with different vocabularies and actions. >>> >>> MGCF/MGW, if needed, will be a natural extension of our >>> CSCF/AS architecture. >>> >>> Just my two cents, to keep the ball rolling, >>> >>> Congratulation again, >>> >>> -giovanni >>> >>> >>> On Tue, Nov 28, 2023 at 2:02 PM Bogdan-Andrei Iancu >>> wrote: >>> >>> Hi all, >>> >>> (disclaimer : cross lists posting is not a good practice >>> - we will do this only to catch the attention and get >>> momentum with this initial topic) >>> >>> As a first step here, is to work out the scope of the >>> IMS implementation in OpenSIPS. IMS is a vast concept, >>> with SIP and non-SIP components, and we want to >>> understand and agree on which components of IMS may be >>> subject of work from the OpenSIPS perspective. For >>> example, we do consider the CSCF as a must here, but we >>> may explore the HSS, AS, MGW or other components. >>> >>> From the OpenSIPS perspective, we look for IMS >>> components which are SIP related. At least as a starting >>> point. So, the first obvious candidate is the *Call >>> Session Control Function (CSCF)*. And here we need to >>> look into and address the specific functionalities of >>> each sub-component: >>>     * P-CSCF >>>     * I-CSCF >>>     * S-CSCF >>> >>> Again, these are the pretty obvious components, still >>> may look into and evaluate (if of an interest of the >>> OpenSIPS IMS implementation) areas as: >>>     * HSS (from interconnection perspective) >>>     * MGCF / MGW  (from interconnection perspective) >>>     * SIP AS >>>     * others ? >>> >>> Any feedback (with explanations and arguments) about >>> what we should consider for our IMS implementation is >>> more the welcome. I set here just a simple starting >>> point, with no limitations or so. Feel free to >>> contribute to the topic >>> >>> >>> Best regards, >>> >>> -- >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> https://www.siphub.com >>> >>> _______________________________________________ >>> Wg-ims mailing list >>> Wg-ims at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >> >> _______________________________________________ >> Wg-ims mailing list >> Wg-ims at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: