[OpenSIPS-Users] Transfer issue
Jeff Kronlage
jeff at data102.com
Thu Nov 12 05:32:26 CET 2009
Anca,
We've given in and started building a front-end/back-end opensips configuration. The 'front end' will be based on our original script and perform the majority of the functions, the 'back end' being the B2BUA that will be involved starting at the initial invite.
Is there any possibility of getting a basic sample Opensips config to go along with the XML config for the B2BUA scenario?
Thanks,
Jeff K
-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
Sent: Tuesday, November 10, 2009 1:43 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Transfer issue
Hi Jeff,
Jeff Kronlage wrote:
> Anca,
>
> Thanks for the quick reply. I tried as you suggested, on a development box, and while it didn't work, it did look a lot more like what I was expecting to see.
>
> However, I guess I am looking for more of a "tack on" solution for what I've already developed. The "right" answer may be to not support REFERs. Is there any way to modify the script scenario to -just- pick up from the REFER and still bridge in the first call leg? It's OK if the answer is 'no', I just need to know what my options are.
>
>
I am sorry, but the answer is really no. The B2BUA must be in the middle
of the call from the beginning.
Regards,
Anca
> Thanks,
>
> Jeff K
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
> Sent: Monday, November 09, 2009 7:17 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Transfer issue
>
> Hi Jeff,
>
>
> Jeff Kronlage wrote:
>
>> Anca,
>>
>> The key pieces of my config file are:
>>
>> Loadmodule "tm.so"
>> loadmodule "b2b_entities.so"
>> loadmodule "b2b_logic.so"
>>
>> modparam("tm", "pass_provisional_replies", 1)
>> modparam("b2b_entities","server_address","sip:opensips at myproxyabc.com")
>> modparam("b2b_logic", "script_scenario", "/usr/local/etc/opensips/refer_script.xml")
>> modparam("b2b_entities", "script_req_route", "b2b_request")
>> modparam("b2b_entities", "script_reply_route", "b2b_reply")
>>
>> ...and down in route[1], just prior to where I would normally call t_relay():
>>
>> if (is_method("REFER")) {
>> b2b_init_request("refer");
>> exit;
>> }
>>
>>
>>
> No, no, no, you should no call b2b_init_request for REFER but for the
> initial INVITE - since the B2BUA must but itself in the middle of the
> call from the beginning.
>
>
>
>> The contents of /usr/local/etc/opensips/refer_script.xml:
>>
>> <?xml version="1.0"?>
>> <scenario id="refer" name="Handle refer at server" param="0" type="script">
>> <init>
>> <bridge>
>> <server>
>> <id>server1</id>
>> </server>
>> <client>
>> <id>client1</id>
>> <type>message</type>
>> <destination>
>> <value type="initial">server1</value>
>> </destination>
>> </client>
>> </bridge>
>> </init>
>>
>> <rules>
>> <request>
>> <refer>
>> <rule id="1">
>> <action>
>> <send_reply>
>> <code>202</code>
>> <reason>Accepted</reason>
>> </send_reply>
>> <end_dialog_leg/>
>> <bridge>
>> <client>
>> <peer/>
>> </client>
>> <client>
>> <id>client2</id>
>> <destination>
>> <value type="header">Refer-To</value>
>> </destination>
>> </client>
>> </bridge>
>> </action>
>> </rule>
>> </refer>
>> </request>
>> </rules>
>> </scenario>
>>
>> So I believe I've done everything you've suggested? The only thing that was a little strange is that when I compiled from the svn, I had to edit /parser/parse_fline.h. I had found items such as INVITE_LEN in there, and my compiler complained that REFER_LEN, as well as several other variables, were undefined. I modified this section of parse_fline.h to the following:
>>
>> #define INVITE "INVITE"
>> #define CANCEL "CANCEL"
>> #define ACK "ACK"
>> #define BYE "BYE"
>> #define INFO "INFO"
>> #define PRACK "PRACK"
>> #define REFER "REFER"
>> #define SUBSCRIBE "SUBSCRIBE"
>> #define NOTIFY "NOTIFY"
>> #define MESSAGE "MESSAGE"
>>
>> #define INVITE_LEN 6
>> #define CANCEL_LEN 6
>> #define ACK_LEN 3
>> #define BYE_LEN 3
>> #define INFO_LEN 4
>> #define PRACK_LEN 5
>> #define REFER_LEN 5
>> #define SUBSCRIBE_LEN 9
>> #define NOTIFY_LEN 6
>> #define MESSAGE_LEN 7
>>
>>
>>
> I forgot to commit the parse_fline.h file on friday. I have commited it
> today. You can delete yours and update from svn.
>
> Regards,
> Anca
>
>> Please advise,
>>
>> Jeff
>>
>> -----Original Message-----
>> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
>> Sent: Monday, November 09, 2009 2:34 AM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Transfer issue
>>
>> Hi Jeff,
>>
>>
>> It seems that the b2b module just does simple forward of REFER request
>> in your setup.
>> Have you loaded the refer scenario? You can find it here:
>> http://www.opensips.org/Resources/B2buaTutorial#toc15. You have to put
>> in in a file and then set the 'script_scenario' parameter to the path of
>> the file:
>>
>> modparam("b2b_logic", "script_scenario", "path_to_scenario_refer.xml")
>>
>> The you have to call the b2b_init_request function with the "refer"
>> parameter:
>> b2b_init_request("refer");
>>
>> Regards,
>> Anca
>>
>>
>> Jeff Kronlage wrote:
>>
>>
>>> Anca,
>>>
>>> Thanks again for your work on this. I've gotten the b2b modules working and I'm attempting to use the REFER scenario, but I'm having some confusion regarding how a REFER with a B2BUA should be handled.
>>>
>>> My test environment looks like this:
>>>
>>> UA1 (softphone) ---INVITE--> [Opensips] ---PSTN_GATEWAY(UA2)--> POTS Phone
>>> ..... (Session progress/OK/etc) ....
>>> UA1 (softphone) ---ACK--> [Opensips] ---> PSTN_GATEWAY(UA2)
>>>
>>> At this point, the first call is setup.
>>>
>>> UA1 (softphone) ---REFER UA2 to UA3--> [Opensips] **b2b module called**
>>> B2B ---REFER B2B to UA3--> PSTN_GATEWAY (UA2)
>>> B2B <--404 NOT FOUND-- PSTN_GATEWAY (UA2)
>>> B2B ---404 NOT FOUND-> UA1 (softphone)
>>>
>>> Obviously this fails. Note this same problem occurs on two completely separate gateways with different hardware. My questions are:
>>>
>>> 1) Shouldn't the b2b scenario send an INVITE off to UA3, wait for the OK/ACK, then simply REFER UA2 to UA3? It seems to me that a B2B -> UA3 refer (which is what I appear to be getting) is out-of-dialog, and many gateways can't start a dialog with a REFER anyway.
>>> 2) If not, any ideas on what I'm doing wrong?
>>>
>>> The pertinent parts of my sip dump are below, beginning with the first REFER, are below:
>>>
>>> REFER sip:18002441111 at 208.94.157.10:5060;transport=udp SIP/2.0
>>> Via: SIP/2.0/UDP 70.57.173.114:63390;branch=z9hG4bK-d8754z-501a700165366947-1---d8754z-;rport
>>> Max-Forwards: 70
>>> Route: <sip:64.111.16.50;lr;ftag=a837e85e;did=847.0253b4b4>
>>> Contact: <sip:7194760273 at 70.57.173.114:63390>
>>> To: <sip:18002441111 at myproxyabc:5060>;tag=b9d5ed0-13c4-4af6e7b5-46585e8b-24ef0427
>>> From: "3CXPhone"<sip:7194760273 at myproxyabc.com:5060>;tag=a837e85e
>>> Call-ID: NDY4YWZhMzRjMDQzZmM2MTU0YTg5YzRlZmFlMzU5NDc.
>>> CSeq: 4 REFER
>>> Proxy-Authorization: <omitted>
>>> User-Agent: 3CXVoipPhone 4.0.9530.0
>>> Refer-To: <sip:18887779569 at prxdev.sip.data102.com:5060>
>>> Referred-By: "3CXPhone"<sip:7194760273 at prxdev.sip.data102.com:5060>
>>> Content-Length: 0
>>>
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 70.57.173.114:63390;branch=z9hG4bK-d8754z-501a700165366947-1---d8754z-;rport=63390
>>> To: <sip:18002441111 at myproxyabc.com:5060>;tag=b9d5ed0-13c4-4af6e7b5-46585e8b-24ef0427
>>> From: "3CXPhone"<sip:7194760273 at myproxyabc.com:5060>;tag=a837e85e
>>> Call-ID: NDY4YWZhMzRjMDQzZmM2MTU0YTg5YzRlZmFlMzU5NDc.
>>> CSeq: 4 REFER
>>> Server: OpenSIPS (1.6.1-notls (i386/linux))
>>> Content-Length: 0
>>>
>>> REFER sip:18002441111 at 208.94.157.10:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 64.111.16.50;branch=z9hG4bK7ea5.271c79b2.0
>>> To: sip:18002441111 at 208.94.157.10:5060
>>> From: <sip:7194760273 at myproxyabc.com:5060>;tag=b9952cfdcb0f3026fcffe5bf74779956-dca7
>>> CSeq: 2 REFER
>>> Call-ID: B2B.462.0.1257695261
>>> Content-Length: 0
>>> User-Agent: OpenSIPS (1.6.1-notls (i386/linux))
>>> Contact: <sip:opensips at myproxyabc.com:5060>
>>>
>>> SIP/2.0 404 Not Found
>>> From: <sip:7194760273 at myproxyabc.com:5060>;tag=b9952cfdcb0f3026fcffe5bf74779956-dca7
>>> To: <sip:18002441111 at 208.94.157.10:5060>;tag=b9d5ed0-13c4-4af6e7c2-465890d5-3ca2cb05
>>> Call-ID: B2B.462.0.1257695261
>>> CSeq: 2 REFER
>>> Via: SIP/2.0/UDP 64.111.16.50;branch=z9hG4bK7ea5.271c79b2.0
>>> Content-Length: 0
>>>
>>> As always, help is much appreciated!
>>>
>>> Jeff K
>>>
>>> -----Original Message-----
>>> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
>>> Sent: Friday, November 06, 2009 9:49 AM
>>> To: OpenSIPS users mailling list
>>> Subject: Re: [OpenSIPS-Users] Transfer issue
>>>
>>> Hi,
>>>
>>> The REFER handing support has been added in B2BUA. Please update from
>>> svn to use this feature.
>>> To enable it you have to load a simple scenario document that describes
>>> the behavior of the B2BUA when a REFER message is received and then call
>>> b2b_init_request("refer") for the initial Invite message.
>>> I have also updated the documentation page and you can find there also
>>> the scenario document for this feature:
>>> http://www.opensips.org/Resources/B2buaTutorial#toc15.
>>>
>>> Regards,
>>> Anca
>>>
>>>
>>> Jeff Kronlage wrote:
>>>
>>>
>>>
>>>> Hi Bogdan,
>>>>
>>>> Thanks for the fantastic news.
>>>>
>>>> I don't suppose you have any samples of how to interpret a REFER and perform a transfer?
>>>>
>>>> I've started pouring through the documentation for the B2BUA, but I'm still grinding through it :)
>>>>
>>>> Thanks,
>>>>
>>>> Jeff
>>>>
>>>> -----Original Message-----
>>>> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
>>>> Sent: Saturday, October 24, 2009 11:18 AM
>>>> To: OpenSIPS users mailling list
>>>> Subject: Re: [OpenSIPS-Users] Transfer issue
>>>>
>>>> Hi Iñaki,
>>>>
>>>> Actually this is why the B2BUA module was designed in opensips - in most
>>>> of the cases you need to control/change the dialog(s) but without any
>>>> media dependencies/penalties (like you have now in most of the IP-PBXs).
>>>>
>>>> So you actually can have a highly scalable signalling B2BUA - the
>>>> opensips module could be used to locally (on opensips) interpret the
>>>> REFER and do the call transfer, totally transparent to the other party.
>>>>
>>>> Regards,
>>>> Bogdan
>>>>
>>>> Iñaki Baz Castillo wrote:
>>>>
>>>>
>>>>
>>>>
>>>>> El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Our setup has been initially
>>>>>> engineered for thousands of concurrent calls, and we're hoping to avoid
>>>>>> having a dozen Asterisk machines :)
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> What you are looking for is the dream all want: a scalable SIP B2BUA (no media
>>>>> handling), so a cluster of these B2BUA's would be located behind a proxy
>>>>> (which does load balancing and failover). And it would be greater if the B2BUA
>>>>> share information (about current dialogs and so) in some way (memcache? common
>>>>> database?...).
>>>>>
>>>>> You could implement it with SipServlets (see Sailin SIP application server or
>>>>> others), or FreeSwitch which allows calls without handling the media...
>>>>> Of course, Asterisk is not the most suitable solution: it involves media
>>>>> handling ("canreinvite" is a hack), it has a very poor SIP stack... and
>>>>> basically it's designed to be a single PBX box.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
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>>>>
>>>>
>>>>
>>>>
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