From faheem2084 at gmail.com  Tue Jan  2 11:24:45 2024
From: faheem2084 at gmail.com (Faheem Muhammad)
Date: Tue, 2 Jan 2024 15:24:45 +0400
Subject: [OpenSIPS-Users] regular expression matching question
Message-ID: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>

Hi everyone,

Firstly, I wish you all a Happy New Year with the best wishes for each
community member and the OpenSIPS project.

I have a question related to regular expressions matching in an IF
condition.
There are two identical regular expressions. One RE is matching and one RE
is not matching the pattern.

*Sample $fU Range:  *+966123456700 to +96612345650

# RE1 - Matching Regular expression
if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
xlog("re matched\n")
}

# RE2  - Non matching Regular expression
if($fU ~="\+9661234567([0-4]\d|50)"){
} # RE not matched

Can anyone please suggest what I'm doing wrong?

*OpenSIPS version*:     3.2.3
OS: *Debian 12*


Best regards,
M. Faheem
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From regardt at rain.co.za  Tue Jan  2 13:27:00 2024
From: regardt at rain.co.za (Regardt van de Vyver)
Date: Tue, 2 Jan 2024 13:27:00 +0000
Subject: [OpenSIPS-Users] Users Digest, Vol 186, Issue 1
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References: <mailman.1.1704196801.3819.users@lists.opensips.org>
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Today's Topics:

   1. regular expression matching question (Faheem Muhammad)


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Message: 1
Date: Tue, 2 Jan 2024 15:24:45 +0400
From: Faheem Muhammad <faheem2084 at gmail.com>
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] regular expression matching question
Message-ID:
        <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hi everyone,

Firstly, I wish you all a Happy New Year with the best wishes for each
community member and the OpenSIPS project.

I have a question related to regular expressions matching in an IF
condition.
There are two identical regular expressions. One RE is matching and one RE
is not matching the pattern.

*Sample $fU Range:  *+966123456700 to +96612345650

# RE1 - Matching Regular expression
if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
xlog("re matched\n")
}

# RE2  - Non matching Regular expression
if($fU ~="\+9661234567([0-4]\d|50)"){
} # RE not matched

Can anyone please suggest what I'm doing wrong?

*OpenSIPS version*:     3.2.3
OS: *Debian 12*


Best regards,
M. Faheem
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From callum.guy at x-on.co.uk  Tue Jan  2 16:27:43 2024
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Tue, 2 Jan 2024 16:27:43 +0000
Subject: [OpenSIPS-Users] regular expression matching question
In-Reply-To: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
References: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
Message-ID: <CAFjCFz=+Mf3YdM5JbBhG7WN1Zqu9Vx705uy6oJVyLSK9jZ-fMw@mail.gmail.com>

Hi Faheem,

This is a simple case of regex support - I believe OpenSIPs uses Extended
Regular Expressions and not Perl Compatible Regular Expressions.

Swap out \d with the longer [0-9] format and your expression should match.

Happy new year,

Callum

On Tue, 2 Jan 2024 at 11:27, Faheem Muhammad <faheem2084 at gmail.com> wrote:

> Hi everyone,
>
> Firstly, I wish you all a Happy New Year with the best wishes for each
> community member and the OpenSIPS project.
>
> I have a question related to regular expressions matching in an IF
> condition.
> There are two identical regular expressions. One RE is matching and one RE
> is not matching the pattern.
>
> *Sample $fU Range:  *+966123456700 to +96612345650
>
> # RE1 - Matching Regular expression
> if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
> xlog("re matched\n")
> }
>
> # RE2  - Non matching Regular expression
> if($fU ~="\+9661234567([0-4]\d|50)"){
> } # RE not matched
>
> Can anyone please suggest what I'm doing wrong?
>
> *OpenSIPS version*:     3.2.3
> OS: *Debian 12*
>
>
> Best regards,
> M. Faheem
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 


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From mcrans at gmail.com  Wed Jan  3 14:33:16 2024
From: mcrans at gmail.com (Michel crans)
Date: Wed, 3 Jan 2024 15:33:16 +0100
Subject: [OpenSIPS-Users] regular expression matching question
In-Reply-To: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
References: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
Message-ID: <CAPWMwMENs-oG7XWxJHMS+Fe16C644CJ5bRivi1=0MDr_j2FJ7w@mail.gmail.com>

The reason why your second regular expression is not matching the pattern
is because of a typo in the syntax. You have used the ~= operator instead
of the =~ operator, which is the correct way to test if a string matches a
regular expression in PCRE.

Op di 2 jan 2024 om 12:28 schreef Faheem Muhammad <faheem2084 at gmail.com>:

> Hi everyone,
>
> Firstly, I wish you all a Happy New Year with the best wishes for each
> community member and the OpenSIPS project.
>
> I have a question related to regular expressions matching in an IF
> condition.
> There are two identical regular expressions. One RE is matching and one RE
> is not matching the pattern.
>
> *Sample $fU Range:  *+966123456700 to +96612345650
>
> # RE1 - Matching Regular expression
> if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
> xlog("re matched\n")
> }
>
> # RE2  - Non matching Regular expression
> if($fU ~="\+9661234567([0-4]\d|50)"){
> } # RE not matched
>
> Can anyone please suggest what I'm doing wrong?
>
> *OpenSIPS version*:     3.2.3
> OS: *Debian 12*
>
>
> Best regards,
> M. Faheem
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From bogdan at opensips.org  Fri Jan  5 10:55:26 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jan 2024 12:55:26 +0200
Subject: [OpenSIPS-Users] uac_replace_from in CANCEL
In-Reply-To: <CALOQYisg-g=6xx=OJ_+1eQHPRE3BFFC9vswYso41wChzLXXZ6g@mail.gmail.com>
References: <CALOQYivKCFZYHHxE_TxUyD4dM22BSPA6Y6h_ORzZxh9UK52t_g@mail.gmail.com>
 <7a5cd98b-4123-40d2-8dd2-32b23131d7bd@opensips.org>
 <CALOQYisg-g=6xx=OJ_+1eQHPRE3BFFC9vswYso41wChzLXXZ6g@mail.gmail.com>
Message-ID: <40f20910-463c-4b7d-9bc5-7aa045b02d75@opensips.org>

Hi Jason,

For changing the FROM hdr (at INVITE time), better use the 
uac_replace_from() function [1] - by doing this, the TM will become 
aware of the change and reflect it into CANCEL too.


[1] 
https://opensips.org/html/docs/modules/3.4.x/uac.html#func_uac_replace_from

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 24.12.2023 13:33, nz deals wrote:
> Hi Bogdan,
> Agreed that the RFC3261 CANCEL is truly built using INVITE. In my 
> case, somehow CANCEL is not the same as the initial INVITE. Due to the 
> complexity of my case I have hard times to fix the issue. Let me 
> explain a bit.
> On the INVITE i change the From Header, which i can see has also been 
> changed during Authorization.
> If i CANCEL then the From header is different (it's the same as i have 
> before making my change).
>
> I am also using topology_hiding().
>
>
> under the main route i change the From header for INVITE
>
>         set_advertised_address("104.13.xx.xx "); # same as i have used 
> in the advertised_address
>         # this is my cancel processing
>         if (is_method("CANCEL")) {
>                 if (t_check_trans())
>                         t_relay(8);
>                 exit;
>         }
>         if (has_totag()) {
>             if(topology_hiding_match())  {
>                                    # remove and add a custom one
>                                    remove_hf("From");
>                                    append_hf("From: 
> \"$fU\"<sip:$fU@$td;custom=$var(custom)>;tag=$ft\r\n");
>  force_send_socket("tls:192.xx.xx.xx:5061");
>              }
>         }
> My opensips version is 3.4.2
> My listening socket is socket=tls:ens192:5061 tag ens192  (its the 
> same 192.xx.xx.xx IP that you see in the force_send_socket)
> I am also using public ip as follows
> advertised_address=104.13.xx.xx
> alias=104.13.xx.xx
>
> Regards,
> Jason
>
> On Thu, 21 Dec 2023 at 02:31, Bogdan-Andrei Iancu 
> <bogdan at opensips.org> wrote:
>
>     Hi Jason,
>
>     In transactional stateful SIP, the CANCEL requests are hop by hop
>     - each hop in the path is generating its own CANCEL requests to
>     the next hop, which consumes it; there is no actual relaying of
>     the CANCELs. So, the replacing (which works in relaying mode only)
>     doesn't fit here.
>
>     Even more, the RFC3261 gives a rigorous way for building the
>     CANCEL requests, they are to be built 100% based on the INVITE
>     request only (nothing more). OpenSIPS internally builds the CANCEL
>     in accordance to the corresponding INVITE, so you should not need
>     any such changes.
>
>     May you detail why you think you need this FROM replacement?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com
>        https://www.siphub.com
>
>     On 20.12.2023 12:38, nz deals wrote:
>>     Hi,
>>     I've been attempting to modify the From header in the CANCEL, but
>>     it seems the changes aren't taking effect. I've also experimented
>>     with remove_hf and append_hf, but unfortunately, these methods
>>     didn't work either. Could someone kindly offer suggestions if I
>>     might be overlooking something?
>>
>>     This is in the main request route
>>
>>             if (is_method("CANCEL")) {
>>                     uac_replace_from("","sip:test at test.com
>>     <mailto:sip%3Atest at test.com>");
>>                     if (t_check_trans())
>>                             t_relay(8);
>>                     exit;
>>             }
>>
>>
>>
>>     Regards,
>>     Jason
>>
>>     _______________________________________________
>>     Users mailing list
>>     Users at lists.opensips.org
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From faheem2084 at gmail.com  Fri Jan  5 11:56:11 2024
From: faheem2084 at gmail.com (Faheem Muhammad)
Date: Fri, 5 Jan 2024 15:56:11 +0400
Subject: [OpenSIPS-Users] regular expression matching question
In-Reply-To: <CAPWMwMENs-oG7XWxJHMS+Fe16C644CJ5bRivi1=0MDr_j2FJ7w@mail.gmail.com>
References: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
 <CAPWMwMENs-oG7XWxJHMS+Fe16C644CJ5bRivi1=0MDr_j2FJ7w@mail.gmail.com>
Message-ID: <CA++nn_xKt7rW5CakGw1LBq5s0iurshZysg1FAjHayJ0JhE8=_A@mail.gmail.com>

Hi Machel,

Appreciate the correction; it was a typo when I posted the question.
However, as Callum mentioned earlier, the "IF" condition with the "=~"
operator doesn't fully support PCRE. It seems that any regular expression
with parentheses isn't matched using the "=~" operator.
Using "pcre_match" or an alternative regular expression would be more
suitable in this case.

Thank you
M. Faheem

On Wed, Jan 3, 2024 at 6:37 PM Michel crans <mcrans at gmail.com> wrote:

>
> The reason why your second regular expression is not matching the pattern
> is because of a typo in the syntax. You have used the ~= operator instead
> of the =~ operator, which is the correct way to test if a string matches
> a regular expression in PCRE.
>
> Op di 2 jan 2024 om 12:28 schreef Faheem Muhammad <faheem2084 at gmail.com>:
>
>> Hi everyone,
>>
>> Firstly, I wish you all a Happy New Year with the best wishes for each
>> community member and the OpenSIPS project.
>>
>> I have a question related to regular expressions matching in an IF
>> condition.
>> There are two identical regular expressions. One RE is matching and one
>> RE is not matching the pattern.
>>
>> *Sample $fU Range:  *+966123456700 to +96612345650
>>
>> # RE1 - Matching Regular expression
>> if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
>> xlog("re matched\n")
>> }
>>
>> # RE2  - Non matching Regular expression
>> if($fU ~="\+9661234567([0-4]\d|50)"){
>> } # RE not matched
>>
>> Can anyone please suggest what I'm doing wrong?
>>
>> *OpenSIPS version*:     3.2.3
>> OS: *Debian 12*
>>
>>
>> Best regards,
>> M. Faheem
>>
>>
>> _______________________________________________
>> 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
>
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From liviu at opensips.org  Mon Jan  8 09:40:13 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 8 Jan 2024 11:40:13 +0200
Subject: [OpenSIPS-Users] mid_register and its possible bad actions with
 De-REGISTER
In-Reply-To: <716391af-fb1a-01a9-e440-34a6e31ec45d@gmail.com>
References: <716391af-fb1a-01a9-e440-34a6e31ec45d@gmail.com>
Message-ID: <a1d0f332-8195-4b62-b51c-6939712898ee@opensips.org>

On 27.12.2023 11:38, Dmitry Ponomaryov wrote:
>
> All this to say, it might make sense to add the ability to disable 
> this De-REGISTER, because with a new registration, we will still send 
> it a registration with an updated contact before the main registrar 
> times out, and there will be no problems, that is, for example, the 
> following settings:
>
> modparam("mid_registrar", "outgoing_expires", 600) 
> modparam("mid_registrar", "default_expires", 300) 

Hello Dmitry,

 From your scenario, it sounds like maybe you should enable AoR 
throttling, not just Ct throttling? Because in Ct throttling, each 
outgoing contact has a unique "ctid=" parameter, so it must be 
de-registered from the main registrar on each contact replace, since the 
replacement Contact has a new "ctid=", making it an entirely different 
SIP URI.

On the other hand, AoR throttling goes beyond this and just makes sure 
there is always 1 AoR registered downstream.  So on a "contact replace" 
operation,  no SIP signaling should reach the Asterisk in that case.

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com
OpenSIPS Summit 2024 Valencia, May 14-17 | www.opensips.org/events



From liviu at opensips.org  Mon Jan  8 10:11:10 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 8 Jan 2024 12:11:10 +0200
Subject: [OpenSIPS-Users] replicate to HEP using the advertised address
 - OPENSIPS 3.3.9
In-Reply-To: <CAAALedoQVm0KNqJX5mWmr1fz-xdem+dXHVu-kSbUrMOmZiWiHg@mail.gmail.com>
References: <CAAALedoQVm0KNqJX5mWmr1fz-xdem+dXHVu-kSbUrMOmZiWiHg@mail.gmail.com>
Message-ID: <54be016c-9ee6-44b7-a6c9-3599e9538c04@opensips.org>

On 26.12.2023 22:54, James Seer wrote:
> Aaaa the good old days before commit 4bd50335a (2023-12-12 by Vlad 
> Paiu), life was simpler when my vip network interface and VIP itself 
> were just one happy entity in HOMER's eyes :P
>
> This recent commit with its replication to HEP using the advertised 
> address has complicated things in homer heplify by treating them as 
> two different entities in the sip sequence (precisely during ACKs 
> CANCEL and BYEs). Any chance we could make this an optional feature?

Hi James,

First of all, VIP issue aside, there seems to be a bug in that patch 
where the /sourceIP/ and /sourcePort/ are changed to /destinationIP /and 
/destinationPort/ without a good reason (currently started a discussion 
with Vlad - we'll get to the bottom of it).  Maybe if this were tweaked, 
your SIP flows would return back to looking normal?!

Nonetheless, if you have a 3.3 source tree to work with, try the 
following patch onto your root sources directory and see if the behavior 
improves by just fixing the bug instead of reverting the patch:

git apply <(base64 -d <<EOF | gzip -dc
H4sIAAAAAAAAA91Ry26DMBA8w1fsqUoCJpCSZ9U0HxD11lMUWcYPZBUwsk3US/+9hiQNOQT13Is9
Hs/ujsdMCgEI5dICmZaKNQU3U6sJ5fqyRRSyRze+rBj/goTN6ex5nq2iKF2ss3XGkiUkcbxIUx8h
9LizHwTBQPfdDtAsSVdhEkNwBY40llhJQVYWjKxxpx8Zqxt6JkqTw8QtIXRXWFZC4RraLeyqCp5j
UZB87IP37YNbDbfYKPqJqSqasjIjYBk+kcIc0mP4i+c9vHBYaFW6eVkjROgjz3Mz0VbTU5S5YDBh
THNj0Jaw07m58xgVvHq7mnW1rQgm46eB0sthc9MwY7GsQ4C/DhzQ1UrbzV3nlglvrWutrBq/+IHn
3RwYTTsHd8SDwuF8l71MVz28dtiqfrpD7x/2/a//pY33GloSHyMHIitL3jLwCi0cvX/s9530B7QY
9NXwAwAA
EOF
)

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com
OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events
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From sj6502621 at gmail.com  Mon Jan  8 10:50:32 2024
From: sj6502621 at gmail.com (James Seer)
Date: Mon, 8 Jan 2024 11:50:32 +0100
Subject: [OpenSIPS-Users] replicate to HEP using the advertised address
 - OPENSIPS 3.3.9
In-Reply-To: <54be016c-9ee6-44b7-a6c9-3599e9538c04@opensips.org>
References: <CAAALedoQVm0KNqJX5mWmr1fz-xdem+dXHVu-kSbUrMOmZiWiHg@mail.gmail.com>
 <54be016c-9ee6-44b7-a6c9-3599e9538c04@opensips.org>
Message-ID: <CAAALedpOcHD_794DqtF1eYt6jt_jSrKoA1cosDKV1z70q0SAsw@mail.gmail.com>

Hi Liviu,

Thanks for the patch. However I can't apply it as my OpenSIPS is installed
as a binary package and not compiled from source.
I've also created an issue on GitHub to track this.
https://github.com/OpenSIPS/opensips/issues/3276

Le lun. 8 janv. 2024 à 11:13, Liviu Chircu <liviu at opensips.org> a écrit :

> On 26.12.2023 22:54, James Seer wrote:
>
> Aaaa the good old days before commit 4bd50335a (2023-12-12 by Vlad Paiu),
> life was simpler when my vip network interface and VIP itself were just one
> happy entity in HOMER's eyes :P
>
> This recent commit with its replication to HEP using the advertised
> address has complicated things in homer heplify by treating them as two
> different entities in the sip sequence (precisely during ACKs CANCEL and
> BYEs). Any chance we could make this an optional feature?
>
> Hi James,
>
> First of all, VIP issue aside, there seems to be a bug in that patch where
> the *sourceIP* and *sourcePort* are changed to *destinationIP *and
> *destinationPort* without a good reason (currently started a discussion
> with Vlad - we'll get to the bottom of it).  Maybe if this were tweaked,
> your SIP flows would return back to looking normal?!
>
> Nonetheless, if you have a 3.3 source tree to work with, try the following
> patch onto your root sources directory and see if the behavior improves by
> just fixing the bug instead of reverting the patch:
>
> git apply <(base64 -d <<EOF | gzip -dc
>
> H4sIAAAAAAAAA91Ry26DMBA8w1fsqUoCJpCSZ9U0HxD11lMUWcYPZBUwsk3US/+9hiQNOQT13Is9
>
> Hs/ujsdMCgEI5dICmZaKNQU3U6sJ5fqyRRSyRze+rBj/goTN6ex5nq2iKF2ss3XGkiUkcbxIUx8h
>
> 9LizHwTBQPfdDtAsSVdhEkNwBY40llhJQVYWjKxxpx8Zqxt6JkqTw8QtIXRXWFZC4RraLeyqCp5j
>
> UZB87IP37YNbDbfYKPqJqSqasjIjYBk+kcIc0mP4i+c9vHBYaFW6eVkjROgjz3Mz0VbTU5S5YDBh
>
> THNj0Jaw07m58xgVvHq7mnW1rQgm46eB0sthc9MwY7GsQ4C/DhzQ1UrbzV3nlglvrWutrBq/+IHn
>
> 3RwYTTsHd8SDwuF8l71MVz28dtiqfrpD7x/2/a//pY33GloSHyMHIitL3jLwCi0cvX/s9530B7QY
> 9NXwAwAA
> EOF
> )
>
> Best regards,
>
> --
> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS Summit 2024 Valencia, May 14-17 | www.opensips.org/events
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From liviu at opensips.org  Mon Jan  8 14:32:13 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 8 Jan 2024 16:32:13 +0200
Subject: [OpenSIPS-Users] uac_auth DB support
In-Reply-To: <CAMtj7Ro2uxqebdNL=QDDp8JwLkOEkS2vPbcvqz8UwWGPQSA-dg@mail.gmail.com>
References: <CAMtj7Ro2uxqebdNL=QDDp8JwLkOEkS2vPbcvqz8UwWGPQSA-dg@mail.gmail.com>
Message-ID: <7338ea31-9828-4d42-b7c0-cda5c0791c2c@opensips.org>

On 06.12.2023 16:18, S.Rosenberg wrote:
>
> I would like to know if OpenSIPS has a way to pull the registrant info 
> via code without manually doing a DB query?

Hi Sebastian,

A bit late to the party here, but try giving the sql_cacher 
<https://opensips.org/docs/modules/3.4.x/sql_cacher.html> a spin.  It 
also works on top of cachedb_local (or any other NoSQL, for that 
matter...), however it will help you:

- cut down on the amount of LoC in your script (just configure the 
cached table properties in the "modparam" section and you're good to go)

- solve the problem of reloading the data (you can configure it with 
on_demand=300, for example, in order to cache a given row for a maximum 
of 5 minutes)

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com
OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events
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From rosenberg11219 at gmail.com  Mon Jan  8 18:42:29 2024
From: rosenberg11219 at gmail.com (Schneur Rosenberg)
Date: Mon, 8 Jan 2024 13:42:29 -0500
Subject: [OpenSIPS-Users] uac_auth DB support
In-Reply-To: <7338ea31-9828-4d42-b7c0-cda5c0791c2c@opensips.org>
References: <CAMtj7Ro2uxqebdNL=QDDp8JwLkOEkS2vPbcvqz8UwWGPQSA-dg@mail.gmail.com>
 <7338ea31-9828-4d42-b7c0-cda5c0791c2c@opensips.org>
Message-ID: <CANvjR0WZYRHDznd65U+AM3VsZ=1inVgX80uAqJk1MPUEUH6WnA@mail.gmail.com>

Thanks Liviu, BTW it's not Sebastian. it's Schneur AKA Scott :-)

On Mon, Jan 8, 2024 at 9:34 AM Liviu Chircu <liviu at opensips.org> wrote:
>
> On 06.12.2023 16:18, S.Rosenberg wrote:
>
>
> I would like to know if OpenSIPS has a way to pull the registrant info via code without manually doing a DB query?
>
> Hi Sebastian,
>
> A bit late to the party here, but try giving the sql_cacher a spin.  It also works on top of cachedb_local (or any other NoSQL, for that matter...), however it will help you:
>
> - cut down on the amount of LoC in your script (just configure the cached table properties in the "modparam" section and you're good to go)
>
> - solve the problem of reloading the data (you can configure it with on_demand=300, for example, in order to cache a given row for a maximum of 5 minutes)
>
> Best regards,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> OpenSIPS Summit 2024 Valencia, May 14-17 | www.opensips.org/events
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From iamhalje at gmail.com  Tue Jan  9 05:15:04 2024
From: iamhalje at gmail.com (Dmitry Ponomaryov)
Date: Tue, 9 Jan 2024 10:15:04 +0500
Subject: [OpenSIPS-Users] mid_register and its possible bad actions with
 De-REGISTER
In-Reply-To: <a1d0f332-8195-4b62-b51c-6939712898ee@opensips.org>
References: <716391af-fb1a-01a9-e440-34a6e31ec45d@gmail.com>
 <a1d0f332-8195-4b62-b51c-6939712898ee@opensips.org>
Message-ID: <1ca1f2c0-1e80-bd67-a11c-b1bffb95a70b@gmail.com>

Hello Liviu, In my case, AoR throttling was used, before issue (1), do 
you also think that this is how it should work? (looks like a CRITICAL) 
 > On the other hand, AoR throttling goes beyond this and just makes 
sure there is always 1 AoR registered downstream. So on a "contact 
replace" operation, no SIP signaling should reach the Asterisk in that 
case. the point is that if different devices are registered under the 
same AoR, one such De-REGISTER, on the final Asterisk completely deletes 
the peer, although in some of the locations there is still data and if 
not for the De-REGISTER, Asterisk would have sent its INVITE, we would 
go through lookup and call all the devices we need from location via 
some route, this is the problem that we don’t always need De-REGISTER if 
there is more than one device under the same AoR(when we don't just 
reduce REGISTER to the main registrar, but use them as REGISTER from 
diff devices), which breaks peers on Asterisk with De-REGISTER.

but in total, it might be worth making such modparam module 
mid_registrar to disable De-REGISTER from the OpenSIPS itself, let the 
registrations themselves expire on the main registrar, and if there is a 
new registration with a new ctid with ct-throttling schema, then we will 
simply update peer again... or if someone needs this feature, then let 
it be possible to choose an interface and not just, for example: 
socket=1.1.1.1:5060 socket=2.2.2.2:5060 in this case De-REGISTER will go 
with 1.1.1.1:5060 on main registrar, because it is simply the first one 
listed socket in cfg. off course, your opinion is needed here, you are 
developer of this module since 2016 ;) Best regards!

(1): https://github.com/OpenSIPS/opensips/issues/3193



On 1/8/24 14:40, Liviu Chircu wrote:
> On 27.12.2023 11:38, Dmitry Ponomaryov wrote:
>>
>> All this to say, it might make sense to add the ability to disable 
>> this De-REGISTER, because with a new registration, we will still send 
>> it a registration with an updated contact before the main registrar 
>> times out, and there will be no problems, that is, for example, the 
>> following settings:
>>
>> modparam("mid_registrar", "outgoing_expires", 600) 
>> modparam("mid_registrar", "default_expires", 300) 
>
> Hello Dmitry,
>
> From your scenario, it sounds like maybe you should enable AoR 
> throttling, not just Ct throttling? Because in Ct throttling, each 
> outgoing contact has a unique "ctid=" parameter, so it must be 
> de-registered from the main registrar on each contact replace, since 
> the replacement Contact has a new "ctid=", making it an entirely 
> different SIP URI.
>
> On the other hand, AoR throttling goes beyond this and just makes sure 
> there is always 1 AoR registered downstream.  So on a "contact 
> replace" operation,  no SIP signaling should reach the Asterisk in 
> that case.
>
> Best regards.


From guillaume.desgeorge at orange.com  Tue Jan  9 09:32:25 2024
From: guillaume.desgeorge at orange.com (guillaume.desgeorge at orange.com)
Date: Tue, 9 Jan 2024 09:32:25 +0000
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
Message-ID: <PAVPR02MB92073CE6871A6D91E6DC52E8836A2@PAVPR02MB9207.eurprd02.prod.outlook.com>

Hi everyone,

We are facing a problem with Opensips 3.3 (but I've done tests on opensips 2.4 and it was the same).

In the dispatcher, if we have a name instead of an IP address in "destination", and that the DNS falls down, we have an error "could not resolve hostname" in the log. Then, Opensips can't resolve the name, there is no more OPTIONS message sent to the destination but the state stays as 0 and if a call arrives, Opensips still route the INVITE to the last known IP address.

I find that's strange :

  *   Either Opensips consider that the DNS is down but the destination may be still up with the last IP address he knows and he could still send OPTIONS messages to the last known IP address
  *   Either Opensips consider that the DNS is down and doesn't send OPTIONS message but in that case he should not send INVITE to the old IP address.

I work with a product that has his own DNS and removes the name of the server in the DNS when the machine is down. So everytime the server is down, Opensips can't resolve the name and still send calls to that server...

Is there any parameters I can use to change that way of working ?

Regards,

    Guillaume
Orange Restricted

____________________________________________________________________________________________________________
Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
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Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
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From alain.bieuzent at free.fr  Tue Jan  9 09:53:37 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 09 Jan 2024 10:53:37 +0100
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
Message-ID: <CE6ABE88-AC2B-49B3-B15B-A305F71FCD6B@free.fr>

Hi Guillaume,

 

have you tried to cache your DNS queries with the dns_cache module (https://opensips.org/html/docs/modules/3.3.x/dns_cache) ?

 

Regards

 

 

De : Users <users-bounces at lists.opensips.org> au nom de <guillaume.desgeorge at orange.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 9 janvier 2024 à 10:35
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : [OpenSIPS-Users] DNS could not resolve hostname in dispatcher

 

Hi everyone,

 

We are facing a problem with Opensips 3.3 (but I’ve done tests on opensips 2.4 and it was the same).

 

In the dispatcher, if we have a name instead of an IP address in “destination”, and that the DNS falls down, we have an error “could not resolve hostname” in the log. Then, Opensips can’t resolve the name, there is no more OPTIONS message sent to the destination but the state stays as 0 and if a call arrives, Opensips still route the INVITE to the last known IP address.

 

I find that’s strange : 
Either Opensips consider that the DNS is down but the destination may be still up with the last IP address he knows and he could still send OPTIONS messages to the last known IP address
Either Opensips consider that the DNS is down and doesn’t send OPTIONS message but in that case he should not send INVITE to the old IP address.
 

I work with a product that has his own DNS and removes the name of the server in the DNS when the machine is down. So everytime the server is down, Opensips can’t resolve the name and still send calls to that server…

 

Is there any parameters I can use to change that way of working ?

 

Regards,

 

    Guillaume

Orange Restricted

 
____________________________________________________________________________________________________________
Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
 
This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
Thank you.
_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users 

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From bullehs at gmail.com  Tue Jan  9 14:40:55 2024
From: bullehs at gmail.com (HS)
Date: Tue, 9 Jan 2024 19:40:55 +0500
Subject: [OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
Message-ID: <CA+PgBoY30WxSCM+JWZh8ze5s9d0XyrGf=5ESiXtRwf96p6rOrQ@mail.gmail.com>

Dear all.

Greetings and a Happy New Year. I want to add a PSTN gateway and the
carrier expects something like: [PREFIX]PHONENUMBER at SBC.ANVEO.COM. I think
I have figured out the [PREFIX] part, but can't seem to add "@sbc.anveo.com"
via the dialplan module. There seems to be a validation error on the
Opensips CP. According to some old documentation it used to be possible to
add something like: sip:\\1 at sbc.anveo.com - but not any more. I have
searched about Opensips/Kamailio, read Flavio's book.

Is it possible to append the correct domain/URI/@sbc.whatever.com to a
phone number dependant on the gateway used? Would this happen in the
config? Dialplan? Dynamic Routing? Dispatcher module?

I am sure this is really basic - references to any tutorials would be
helpful.

Thanks.
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From rvg at rvgeerligs.nl  Tue Jan  9 18:30:26 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Tue, 09 Jan 2024 18:30:26 +0000
Subject: [OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
In-Reply-To: <CA+PgBoY30WxSCM+JWZh8ze5s9d0XyrGf=5ESiXtRwf96p6rOrQ@mail.gmail.com>
References: <CA+PgBoY30WxSCM+JWZh8ze5s9d0XyrGf=5ESiXtRwf96p6rOrQ@mail.gmail.com>
Message-ID: <1ab6b1b32f2f650ea584f7d68fed5834535d8737@rvgeerligs.nl>

Hi,

Shouldn't this be done in dynamic routing > Gateway | Carriers?



Regards,


Ronald Geerligs

PS I registered at anveo for an account too.

January 9, 2024 at 11:40 AM, "HS" <bullehs at gmail.com> wrote:

> 
> Dear all.
> 
> Greetings and a Happy New Year. I want to add a PSTN gateway and the carrier expects something like: [PREFIX]PHONENUMBER at SBC.ANVEO.COM. I think I have figured out the [PREFIX] part, but can't seem to add "@sbc.anveo.com http://sbc.anveo.com/ " via the dialplan module. There seems to be a validation error on the Opensips CP. According to some old documentation it used to be possible to add something like: sip:\\1 at sbc.anveo.com - but not any more. I have searched about Opensips/Kamailio, read Flavio's book.
> 
> Is it possible to append the correct domain/URI/@sbc.whatever.com http://sbc.whatever.com/  to a phone number dependant on the gateway used? Would this happen in the config? Dialplan? Dynamic Routing? Dispatcher module?
> 
> I am sure this is really basic - references to any tutorials would be helpful.
> 
> Thanks.
>
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From yonge.liu at moz.com  Wed Jan 10 15:46:34 2024
From: yonge.liu at moz.com (Yonge Liu)
Date: Wed, 10 Jan 2024 10:46:34 -0500
Subject: [OpenSIPS-Users] SIPssert test tool tryout
Message-ID: <CADh3yC487Ffngq+JEgGTr3zAVffpsGAQwLd7OFwA7r61xfv48A@mail.gmail.com>

Hi All,

I am trying to experiment with SIPssert test tool.
By following the guide here: https://github.com/OpenSIPS/SIPssert, I am
having issue to run the tool:

ubuntu:~$ *sipssert /home/ubuntu/tests/opensips/*
Traceback (most recent call last):
  File "/usr/lib/python3/dist-packages/urllib3/connectionpool.py", line
700, in urlopen
    httplib_response = self._make_request(
  File "/usr/lib/python3/dist-packages/urllib3/connectionpool.py", line
395, in _make_request
    conn.request(method, url, **httplib_request_kw)
  File "/usr/lib/python3.10/http/client.py", line 1283, in request
    self._send_request(method, url, body, headers, encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1329, in _send_request
    self.endheaders(body, encode_chunked=encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1278, in endheaders
    self._send_output(message_body, encode_chunked=encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1038, in _send_output
    self.send(msg)
  File "/usr/lib/python3.10/http/client.py", line 976, in send
    self.connect()
  File "/usr/lib/python3/dist-packages/docker/transport/unixconn.py", line
30, in connect
    sock.connect(self.unix_socket)
FileNotFoundError: [Errno 2] No such file or directory

During handling of the above exception, another exception occurred:

Traceback (most recent call last):
  File "/usr/lib/python3/dist-packages/requests/adapters.py", line 439, in
send
    resp = conn.urlopen(
  File "/usr/lib/python3/dist-packages/urllib3/connectionpool.py", line
756, in urlopen
    retries = retries.increment(
  File "/usr/lib/python3/dist-packages/urllib3/util/retry.py", line 532, in
increment
    raise six.reraise(type(error), error, _stacktrace)
  File "/usr/lib/python3/dist-packages/six.py", line 718, in reraise
    raise value.with_traceback(tb)
  File "/usr/lib/python3/dist-packages/urllib3/connectionpool.py", line
700, in urlopen
    httplib_response = self._make_request(
  File "/usr/lib/python3/dist-packages/urllib3/connectionpool.py", line
395, in _make_request
    conn.request(method, url, **httplib_request_kw)
  File "/usr/lib/python3.10/http/client.py", line 1283, in request
    self._send_request(method, url, body, headers, encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1329, in _send_request
    self.endheaders(body, encode_chunked=encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1278, in endheaders
    self._send_output(message_body, encode_chunked=encode_chunked)
  File "/usr/lib/python3.10/http/client.py", line 1038, in _send_output
    self.send(msg)
  File "/usr/lib/python3.10/http/client.py", line 976, in send
    self.connect()
  File "/usr/lib/python3/dist-packages/docker/transport/unixconn.py", line
30, in connect
    sock.connect(self.unix_socket)
urllib3.exceptions.ProtocolError: ('Connection aborted.',
FileNotFoundError(2, 'No such file or directory'))

During handling of the above exception, another exception occurred:

Traceback (most recent call last):
  File "/usr/lib/python3/dist-packages/docker/api/client.py", line 214, in
_retrieve_server_version
    return self.version(api_version=False)["ApiVersion"]
  File "/usr/lib/python3/dist-packages/docker/api/daemon.py", line 181, in
version
    return self._result(self._get(url), json=True)
  File "/usr/lib/python3/dist-packages/docker/utils/decorators.py", line
46, in inner
    return f(self, *args, **kwargs)
  File "/usr/lib/python3/dist-packages/docker/api/client.py", line 237, in
_get
    return self.get(url, **self._set_request_timeout(kwargs))
  File "/usr/lib/python3/dist-packages/requests/sessions.py", line 557, in
get
    return self.request('GET', url, **kwargs)
  File "/usr/lib/python3/dist-packages/requests/sessions.py", line 544, in
request
    resp = self.send(prep, **send_kwargs)
  File "/usr/lib/python3/dist-packages/requests/sessions.py", line 657, in
send
    r = adapter.send(request, **kwargs)
  File "/usr/lib/python3/dist-packages/requests/adapters.py", line 498, in
send
    raise ConnectionError(err, request=request)
requests.exceptions.ConnectionError: ('Connection aborted.',
FileNotFoundError(2, 'No such file or directory'))

During handling of the above exception, another exception occurred:

Traceback (most recent call last):
  File "/usr/local/bin/sipssert", line 4, in <module>
    __import__('pkg_resources').run_script('sipssert==0.1.0', 'sipssert')
  File "/usr/lib/python3/dist-packages/pkg_resources/__init__.py", line
656, in run_script
    self.require(requires)[0].run_script(script_name, ns)
  File "/usr/lib/python3/dist-packages/pkg_resources/__init__.py", line
1453, in run_script
    exec(code, namespace, namespace)
  File
"/usr/local/lib/python3.10/dist-packages/sipssert-0.1.0-py3.10.egg/EGG-INFO/scripts/sipssert",
line 9, in <module>
    run_test_sipssert()
  File
"/usr/local/lib/python3.10/dist-packages/sipssert-0.1.0-py3.10.egg/EGG-INFO/scripts/sipssert",
line 6, in run_test_sipssert
    main.main()
  File
"/usr/local/lib/python3.10/dist-packages/sipssert-0.1.0-py3.10.egg/sipssert/main.py",
line 77, in main
    ctrl = controller.Controller(args)
  File
"/usr/local/lib/python3.10/dist-packages/sipssert-0.1.0-py3.10.egg/sipssert/controller.py",
line 56, in __init__
    self.docker = docker.from_env()
  File "/usr/lib/python3/dist-packages/docker/client.py", line 96, in
from_env
    return cls(
  File "/usr/lib/python3/dist-packages/docker/client.py", line 45, in
__init__
    self.api = APIClient(*args, **kwargs)
  File "/usr/lib/python3/dist-packages/docker/api/client.py", line 197, in
__init__
    self._version = self._retrieve_server_version()
  File "/usr/lib/python3/dist-packages/docker/api/client.py", line 221, in
_retrieve_server_version
    raise DockerException(
docker.errors.DockerException: Error while fetching server API version:
('Connection aborted.', FileNotFoundError(2, 'No such file or directory'))

Attachments are the files in /home/ubuntu/tests/opensips/. Not sure what I
am missing. Appreciate any guide to make this tool work.

Thank you very much,

Yonge

-- 


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From alain.bieuzent at free.fr  Wed Jan 10 15:55:30 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Wed, 10 Jan 2024 16:55:30 +0100
Subject: [OpenSIPS-Users] dispatcher ds_list "full" parameter
Message-ID: <49AC253F-BB9C-4034-B201-187EAD9D14BC@free.fr>

Hi All,

 

On dispatcher module, exported mi function ds_list can have 2 optional parameters (full and partition)

 

When I tried to use these parameters with opensips-cli, it returns an error message :

 

root at lbsip-test:~# opensips-cli -x mi ds_list full

ERROR: command 'ds_list' returned: -32602: Invalid params (Ambiguous call, use named parameters instead)

 

I didn't understand the documentation or is there a bug?

 

PS : I’m using opensips 3.3.6

Regards

 

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From bogdan at opensips.org  Wed Jan 10 16:05:52 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Jan 2024 18:05:52 +0200
Subject: [OpenSIPS-Users] dispatcher ds_list "full" parameter
In-Reply-To: <49AC253F-BB9C-4034-B201-187EAD9D14BC@free.fr>
References: <49AC253F-BB9C-4034-B201-187EAD9D14BC@free.fr>
Message-ID: <368a2391-aa39-43fe-8841-aae757ff5f31@opensips.org>

Hi Alain,

As both parameters (partition and full) are optional, you need to pass 
the name and value of the param (if only one param is passed), otherwise 
OpenSIPS will have no idea if the value belongs to partition or full

Try `opensips-cli -x mi ds_list full=1` , where "full" is the name of 
the param and "1" is its value.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 10.01.2024 17:55, Alain Bieuzent wrote:
>
> Hi All,
>
> On dispatcher module, exported mi function ds_list can have 2 optional 
> parameters (full and partition)
>
> When I tried to use these parameters with opensips-cli, it returns an 
> error message :
>
> root at lbsip-test:~# opensips-cli -x mi ds_list full
>
> ERROR: command 'ds_list' returned: -32602: Invalid params (Ambiguous 
> call, use named parameters instead)
>
> I didn't understand the documentation or is there a bug?
>
> PS : I’m using opensips 3.3.6
>
> Regards
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From bogdan at opensips.org  Wed Jan 10 16:09:42 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Jan 2024 18:09:42 +0200
Subject: [OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
In-Reply-To: <CA+PgBoY30WxSCM+JWZh8ze5s9d0XyrGf=5ESiXtRwf96p6rOrQ@mail.gmail.com>
References: <CA+PgBoY30WxSCM+JWZh8ze5s9d0XyrGf=5ESiXtRwf96p6rOrQ@mail.gmail.com>
Message-ID: <b7aab0c4-6e84-425a-a740-55051b7d836d@opensips.org>

Hi and A Happy New Year too,

IF you use the drouting module for routing to GWs, the module will 
automatically replace the host part of the RURI with the address of the 
GW you configured in the Drouting module.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 09.01.2024 16:40, HS wrote:
> Dear all.
>
> Greetings and a Happy New Year. I want to add a PSTN gateway and the 
> carrier expects something like: [PREFIX]PHONENUMBER at SBC.ANVEO.COM. I 
> think I have figured out the [PREFIX] part, but can't seem to add 
> "@sbc.anveo.com <http://sbc.anveo.com>" via the dialplan module. There 
> seems to be a validation error on the Opensips CP. According to some 
> old documentation it used to be possible to add something like: 
> sip:\\1 at sbc.anveo.com - but not any more. I have searched about 
> Opensips/Kamailio, read Flavio's book.
>
> Is it possible to append the correct domain/URI/@sbc.whatever.com 
> <http://sbc.whatever.com> to a phone number dependant on the gateway 
> used? Would this happen in the config? Dialplan? Dynamic Routing? 
> Dispatcher module?
>
> I am sure this is really basic - references to any tutorials would be 
> helpful.
>
> Thanks.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From alain.bieuzent at free.fr  Wed Jan 10 16:16:19 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Wed, 10 Jan 2024 17:16:19 +0100
Subject: [OpenSIPS-Users] dispatcher ds_list "full" parameter
In-Reply-To: <368a2391-aa39-43fe-8841-aae757ff5f31@opensips.org>
References: <49AC253F-BB9C-4034-B201-187EAD9D14BC@free.fr>
 <368a2391-aa39-43fe-8841-aae757ff5f31@opensips.org>
Message-ID: <B63F3008-FDEA-40F6-934F-CC306415BD89@free.fr>

Thanks bogdan

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : mercredi 10 janvier 2024 à 17:06
À : OpenSIPS users mailling list <users at lists.opensips.org>, Alain Bieuzent <alain.bieuzent at free.fr>
Objet : Re: [OpenSIPS-Users] dispatcher ds_list "full" parameter

 

Hi Alain,

As both parameters (partition and full) are optional, you need to pass the name and value of the param (if only one param is passed), otherwise OpenSIPS will have no idea if the value belongs to partition or full

Try `opensips-cli -x mi ds_list full=1` , where "full" is the name of the param and "1" is its value.

Regards,

Bogdan-Andrei Iancu
 
OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
On 10.01.2024 17:55, Alain Bieuzent wrote:

Hi All,

 

On dispatcher module, exported mi function ds_list can have 2 optional parameters (full and partition)

 

When I tried to use these parameters with opensips-cli, it returns an error message :

 

root at lbsip-test:~# opensips-cli -x mi ds_list full

ERROR: command 'ds_list' returned: -32602: Invalid params (Ambiguous call, use named parameters instead)

 

I didn't understand the documentation or is there a bug?

 

PS : I’m using opensips 3.3.6

Regards

 



_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



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From bogdan at opensips.org  Wed Jan 10 16:27:36 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 10 Jan 2024 18:27:36 +0200
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
In-Reply-To: <PAVPR02MB92073CE6871A6D91E6DC52E8836A2@PAVPR02MB9207.eurprd02.prod.outlook.com>
References: <PAVPR02MB92073CE6871A6D91E6DC52E8836A2@PAVPR02MB9207.eurprd02.prod.outlook.com>
Message-ID: <50525c12-d3f9-4f3a-98d6-c8478248dd7b@opensips.org>

Hi Guillaume,

Indeed, a weird one, not marking the destination as failed upon failure 
to ping it. I checked the drouting module and it is doing such disable, 
so it might be a overlook in the dispatcher module. Please open a bug 
report on the github tracker, so we can address this issue.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 09.01.2024 11:32, guillaume.desgeorge at orange.com wrote:
>
> Hi everyone,
>
> We are facing a problem with Opensips 3.3 (but I’ve done tests on 
> opensips 2.4 and it was the same).
>
> In the dispatcher, if we have a name instead of an IP address in 
> “destination”, and that the DNS falls down, we have an error “could 
> not resolve hostname” in the log. Then, Opensips can’t resolve the 
> name, there is no more OPTIONS message sent to the destination but the 
> state stays as 0 and if a call arrives, Opensips still route the 
> INVITE to the last known IP address.
>
> I find that’s strange :
>
>   * Either Opensips consider that the DNS is down but the destination
>     may be still up with the last IP address he knows and he could
>     still send OPTIONS messages to the last known IP address
>   * Either Opensips consider that the DNS is down and doesn’t send
>     OPTIONS message but in that case he should not send INVITE to the
>     old IP address.
>
> I work with a product that has his own DNS and removes the name of the 
> server in the DNS when the machine is down. So everytime the server is 
> down, Opensips can’t resolve the name and still send calls to that server…
>
> Is there any parameters I can use to change that way of working ?
>
> Regards,
>
>     Guillaume
>
> Orange Restricted
>
> ____________________________________________________________________________________________________________
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged information that may be protected by law;
> they should not be distributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and delete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
> Thank you.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From trevor at webon.co.za  Thu Jan 11 13:55:06 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Thu, 11 Jan 2024 15:55:06 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
Message-ID: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>

Hi All, 

I have a simple issue I need to solve but cant seem to fine a elegant
way of resolving.

I am using acc module + rate_cacher to do some simple billing.

I use $DLG_lifetime but this value seems to be a rounded to the nearest
second what I need is to round up to the nearest second similar to how
ACC module calculates duration

Here is an example CDR


*************************** 1. row ***************************
       duration: 35
    ms_duration: 34168
billed_duration: 34

As you can see duration is rounded up and billed_duration is rounded
down.

Is there any way to get access to the ms of the dialogue from any
module, I could resort to initializing my own timers but there is
already a timer running in the acc module and would be more efficient
to just use it.


Here is how I am creating billed_duration

if (has_totag()) {
    ....
    ....
    ....
    if (is_method("BYE")) {
        if ($DLG_lifetime == 0) {
            $acc_extra(billed_duration) = 1;
        }else{
            $acc_extra(billed_duration) = $DLG_lifetime;
        }
    }

    route(RELAY);
    exit;
}


Thanks
Trevor Steyn




From bullehs at gmail.com  Fri Jan 12 09:37:16 2024
From: bullehs at gmail.com (HS)
Date: Fri, 12 Jan 2024 14:37:16 +0500
Subject: [OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
Message-ID: <CA+PgBoafHE2yJ8=5Ayq=oYcrV8J1Sgwi=veTDNkpmg7+M7cNKg@mail.gmail.com>

Hi all.

Thanks a lot @Bogdan-Andrei Iancu <bogdan at opensips.org>, once you confirmed
that it appends the correct domain, I had to play around with the dialplan
and config and made it work.

@Ronald - thanks. Hope it works for you too.

Best.
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From guillaume.desgeorge at orange.com  Fri Jan 12 09:56:15 2024
From: guillaume.desgeorge at orange.com (guillaume.desgeorge at orange.com)
Date: Fri, 12 Jan 2024 09:56:15 +0000
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
In-Reply-To: <CE6ABE88-AC2B-49B3-B15B-A305F71FCD6B@free.fr>
References: <CE6ABE88-AC2B-49B3-B15B-A305F71FCD6B@free.fr>
Message-ID: <PAVPR02MB9207FE846B92A7A3E580EE5B836F2@PAVPR02MB9207.eurprd02.prod.outlook.com>

Hi Alain,

Thank you for your help and your suggestion.

We are trying to implement DNS_CACHE module with a record "oms" in etc/hosts (same in dispatcher) and Opensips doesn't resolve the name when DNS_CACHE is activated.

Here are our declarations in opensips.cfg file :

### DNS CACHE MODULE ###loadmodule "cachedb_local.so"
loadmodule "cachedb_sql.so"
loadmodule "dns_cache.so"
modparam("cachedb_sql", "cachedb_url", "sql:mysql://opensips:opensipsrw at ossdb/opensips")
modparam("cachedb_sql", "db_table","cachedb")
modparam("dns_cache", "cachedb_url", "sql:mysql://opensips:opensipsrw at ossdb/opensips")

Here are the logs :
CRITICAL:core:mk_proxy: could not resolve hostname: "oms"
Jan 12 10:47:52 [37] ERROR:dispatcher:add_dest2list: could not resolve oms, skipping it
Jan 12 10:47:52 [37] WARNING:dispatcher:ds_load_data: failed to add destination sip:oms:5060 in group 100
Jan 12 10:47:52 [37] WARNING:dispatcher:ds_load_data: No record loaded from db, running on empty sets

There is another name ("rtpe") declared in etc/hosts is also used in opensips.cfg and there is no problem to resolve it. The problem seams to be only in dispatcher.

Are we missing something in our way of using it ?

Regards,

   Guillaume

De : Users <users-bounces at lists.opensips.org> De la part de Alain Bieuzent
Envoyé : mardi 9 janvier 2024 10:54
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher

Hi Guillaume,

have you tried to cache your DNS queries with the dns_cache module (https://opensips.org/html/docs/modules/3.3.x/dns_cache) ?

Regards


De : Users <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> au nom de <guillaume.desgeorge at orange.com<mailto:guillaume.desgeorge at orange.com>>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Date : mardi 9 janvier 2024 à 10:35
À : OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Objet : [OpenSIPS-Users] DNS could not resolve hostname in dispatcher

Hi everyone,

We are facing a problem with Opensips 3.3 (but I've done tests on opensips 2.4 and it was the same).

In the dispatcher, if we have a name instead of an IP address in "destination", and that the DNS falls down, we have an error "could not resolve hostname" in the log. Then, Opensips can't resolve the name, there is no more OPTIONS message sent to the destination but the state stays as 0 and if a call arrives, Opensips still route the INVITE to the last known IP address.

I find that's strange :

  *   Either Opensips consider that the DNS is down but the destination may be still up with the last IP address he knows and he could still send OPTIONS messages to the last known IP address
  *   Either Opensips consider that the DNS is down and doesn't send OPTIONS message but in that case he should not send INVITE to the old IP address.

I work with a product that has his own DNS and removes the name of the server in the DNS when the machine is down. So everytime the server is down, Opensips can't resolve the name and still send calls to that server...

Is there any parameters I can use to change that way of working ?

Regards,

    Guillaume
Orange Restricted


____________________________________________________________________________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc

pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler

a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,

Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.



This message and its attachments may contain confidential or privileged information that may be protected by law;

they should not be distributed, used or copied without authorisation.

If you have received this email in error, please notify the sender and delete this message and its attachments.

As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.

Thank you.
_______________________________________________ Users mailing list Users at lists.opensips.org<mailto:Users at lists.opensips.org> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
____________________________________________________________________________________________________________
Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
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From rvg at rvgeerligs.nl  Sat Jan 13 00:56:42 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Sat, 13 Jan 2024 00:56:42 +0000
Subject: [OpenSIPS-Users] Adding PSTN Gateway in Opensips 3.0
In-Reply-To: <CA+PgBoafHE2yJ8=5Ayq=oYcrV8J1Sgwi=veTDNkpmg7+M7cNKg@mail.gmail.com>
References: <CA+PgBoafHE2yJ8=5Ayq=oYcrV8J1Sgwi=veTDNkpmg7+M7cNKg@mail.gmail.com>
Message-ID: <9b78647377ddaae3d4850afc3e8e9218e339d07a@rvgeerligs.nl>

Thx,

But I have it working allready and I have a choise of PSTN gateways too.

Regards



January 12, 2024 at 6:37 AM, "HS" <bullehs at gmail.com> wrote:


> 
> Hi all.
> 
> Thanks a lot @Bogdan-Andrei Iancu, once you confirmed that it appends the correct domain, I had to play around with the dialplan and config and made it work.
> 
> @Ronald - thanks. Hope it works for you too.
> 
> Best.
>
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From rvg at rvgeerligs.nl  Sat Jan 13 20:37:52 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Sat, 13 Jan 2024 20:37:52 +0000
Subject: [OpenSIPS-Users] pn push notifications
Message-ID: <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>

Hi All,


I am trying to implement an opensips server 3.4 with pn support. Primarily apple push notifications. Later I will try google.

Does any one have this implemented yet?

Where can I find an example cfg for opensips?

Any help is appreciated.


Regards,


Ronald Geerligs
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From igorolhovskiy at gmail.com  Sun Jan 14 17:00:31 2024
From: igorolhovskiy at gmail.com (Ihor Olkhovskyi)
Date: Sun, 14 Jan 2024 18:00:31 +0100
Subject: [OpenSIPS-Users] Removing header by contents
Message-ID: <90b1cb29-f628-4480-b789-155897126d0c@gmail.com>

Hello!

Is there any easy possibility to remove header not by name, but name and 
contents or just header number?

I have several Route headers and want to remove the first one only.

For the moment I'm thinking to remove all headers and add all but the 
first in the loop, but maybe it's an overkill?

P.S.: Why I'm doing this - trying to implement own loose_route for some 
experiments

Thanks in advance,
Ihor



From ag at ag-projects.com  Sun Jan 14 18:13:47 2024
From: ag at ag-projects.com (Adrian Georgescu)
Date: Sun, 14 Jan 2024 15:13:47 -0300
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>
References: <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>
Message-ID: <2FE28EAA-BFF2-4913-A19B-BED2789B79FF@ag-projects.com>

There are many ways of implementing push notifications with OpenSIPS. This is how we implemented it.

https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg

—
Adrian




> On 13 Jan 2024, at 17:37, rvg at rvgeerligs.nl wrote:
> 
> Hi All,
> 
> 
> I am trying to implement an opensips server 3.4 with pn support. Primarily apple push notifications. Later I will try google.
> 
> Does any one have this implemented yet?
> 
> Where can I find an example cfg for opensips?
> 
> Any help is appreciated.
> 
> 
> Regards,
> 
> 
> Ronald Geerligs
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From rvg at rvgeerligs.nl  Sun Jan 14 23:33:32 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Sun, 14 Jan 2024 23:33:32 +0000
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <2FE28EAA-BFF2-4913-A19B-BED2789B79FF@ag-projects.com>
References: <2FE28EAA-BFF2-4913-A19B-BED2789B79FF@ag-projects.com>
 <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>
Message-ID: <53b34a75725ab74eede1afe8b4996afa940729c7@rvgeerligs.nl>

Thank you very much, Adrian, 

I noticed it allready. It seems this is a dedicated push server.
Does it handle the normal SIP traffic as well?

Regards,

Ronald


January 14, 2024 at 3:13 PM, "Adrian Georgescu" <ag at ag-projects.com> wrote:


> 
> There are many ways of implementing push notifications with OpenSIPS. This is how we implemented it.
> 
> https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg
> 
> —
> Adrian
> 
> > 
> > On 13 Jan 2024, at 17:37, rvg at rvgeerligs.nl wrote:
> >  
> >  Hi All,
> >  
> >  
> >  I am trying to implement an opensips server 3.4 with pn support. Primarily apple push notifications. Later I will try google.
> >  
> >  Does any one have this implemented yet?
> >  
> >  Where can I find an example cfg for opensips?
> >  
> >  Any help is appreciated.
> >  
> >  
> >  Regards,
> >  
> >  
> >  Ronald Geerligs
> >  _______________________________________________
> >  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
>
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From guillaume.desgeorge at orange.com  Mon Jan 15 08:19:34 2024
From: guillaume.desgeorge at orange.com (guillaume.desgeorge at orange.com)
Date: Mon, 15 Jan 2024 08:19:34 +0000
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
In-Reply-To: <50525c12-d3f9-4f3a-98d6-c8478248dd7b@opensips.org>
References: <PAVPR02MB92073CE6871A6D91E6DC52E8836A2@PAVPR02MB9207.eurprd02.prod.outlook.com>
 <50525c12-d3f9-4f3a-98d6-c8478248dd7b@opensips.org>
Message-ID: <PAVPR02MB92071B212CD91AABEB4C9684836C2@PAVPR02MB9207.eurprd02.prod.outlook.com>

Hi Bogdan,
Thanks for your answer.
Issue opened : #3285
Regards,

  Guillaume Desgeorge


De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Envoyé : mercredi 10 janvier 2024 17:28
À : OpenSIPS users mailling list <users at lists.opensips.org>; DESGEORGE Guillaume INNOV/IT-S <guillaume.desgeorge at orange.com>
Objet : Re: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher

Hi Guillaume,

Indeed, a weird one, not marking the destination as failed upon failure to ping it. I checked the drouting module and it is doing such disable, so it might be a overlook in the dispatcher module. Please open a bug report on the github tracker, so we can address this issue.

Regards,


Bogdan-Andrei Iancu



OpenSIPS Founder and Developer

  https://www.opensips-solutions.com<https://www.opensips-solutions.com/>

  https://www.siphub.com<https://www.siphub.com/>
On 09.01.2024 11:32, guillaume.desgeorge at orange.com<mailto:guillaume.desgeorge at orange.com> wrote:
Hi everyone,

We are facing a problem with Opensips 3.3 (but I've done tests on opensips 2.4 and it was the same).

In the dispatcher, if we have a name instead of an IP address in "destination", and that the DNS falls down, we have an error "could not resolve hostname" in the log. Then, Opensips can't resolve the name, there is no more OPTIONS message sent to the destination but the state stays as 0 and if a call arrives, Opensips still route the INVITE to the last known IP address.

I find that's strange :

  1.  Either Opensips consider that the DNS is down but the destination may be still up with the last IP address he knows and he could still send OPTIONS messages to the last known IP address
  2.  Either Opensips consider that the DNS is down and doesn't send OPTIONS message but in that case he should not send INVITE to the old IP address.

I work with a product that has his own DNS and removes the name of the server in the DNS when the machine is down. So everytime the server is down, Opensips can't resolve the name and still send calls to that server...

Is there any parameters I can use to change that way of working ?

Regards,

    Guillaume
Orange Restricted


____________________________________________________________________________________________________________

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Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.



This message and its attachments may contain confidential or privileged information that may be protected by law;

they should not be distributed, used or copied without authorisation.

If you have received this email in error, please notify the sender and delete this message and its attachments.

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Thank you.



_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

____________________________________________________________________________________________________________
Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.

This message and its attachments may contain confidential or privileged information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
Thank you.
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From ag at ag-projects.com  Mon Jan 15 11:59:55 2024
From: ag at ag-projects.com (Adrian Georgescu)
Date: Mon, 15 Jan 2024 08:59:55 -0300
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <53b34a75725ab74eede1afe8b4996afa940729c7@rvgeerligs.nl>
References: <2FE28EAA-BFF2-4913-A19B-BED2789B79FF@ag-projects.com>
 <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>
 <53b34a75725ab74eede1afe8b4996afa940729c7@rvgeerligs.nl>
Message-ID: <6C0F9AFC-196F-40B9-9E30-3ECB53DB65FF@ag-projects.com>

OpenSIPS handles the SIP traffic, of course. The push server just handles multiple combinations of apps/ push services in an easier way. You could do a lot or all that its needed just from within OpenSIPS but as the number of combinations of apps and services grow another layer of abstraction make it easier to manage, plus you eliminate another possible blocking item from your routes as the works are down asynchronously outside OpenSIPS.
 
—
Adrian




> On 14 Jan 2024, at 20:33, rvg at rvgeerligs.nl wrote:
> 
> Thank you very much, Adrian,
> 
> I noticed it allready. It seems this is a dedicated push server.
> Does it handle the normal SIP traffic as well?
> 
> Regards,
> 
> Ronald
> January 14, 2024 at 3:13 PM, "Adrian Georgescu" <ag at ag-projects.com <mailto:ag at ag-projects.com?to=%22Adrian%20Georgescu%22%20%3Cag%40ag-projects.com%3E>> wrote:
> 
> There are many ways of implementing push notifications with OpenSIPS. This is how we implemented it.
> 
> https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg
> 
> —
> Adrian
> 
> 
> 
> 
> On 13 Jan 2024, at 17:37, rvg at rvgeerligs.nl <mailto:rvg at rvgeerligs.nl> wrote:
> Hi All,
> I am trying to implement an opensips server 3.4 with pn support. Primarily apple push notifications. Later I will try google.
> Does any one have this implemented yet?
> Where can I find an example cfg for opensips?
> Any help is appreciated.
> Regards,
> Ronald Geerligs
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto: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

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From slackway2me at gmail.com  Mon Jan 15 13:57:29 2024
From: slackway2me at gmail.com (Alexey)
Date: Mon, 15 Jan 2024 18:57:29 +0500
Subject: [OpenSIPS-Users] pn push notifications
Message-ID: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>

Hello,

there are two tutorials in the official blog:

part I :  https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/

part II : https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/

-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From slackway2me at gmail.com  Mon Jan 15 14:03:23 2024
From: slackway2me at gmail.com (Alexey)
Date: Mon, 15 Jan 2024 19:03:23 +0500
Subject: [OpenSIPS-Users] Removing header by contents
Message-ID: <CAPc6LUHPmDwOyFQ+fF2wFVCSvnmku9Mzx9QqxdBXoKuz-ADUZg@mail.gmail.com>

Hello Igor,

these functions - isn't it what you need?

remove_hf_re(hname_expr) [1]
remove_hf_glob(hname_pattern) [2]

1. https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_re
2. https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_glob


-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From rvg at rvgeerligs.nl  Mon Jan 15 16:21:28 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Mon, 15 Jan 2024 16:21:28 +0000
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>
References: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>
Message-ID: <4e85380126b03fe7be85b4027f7342d3ded9e0e9@rvgeerligs.nl>

Hi Alexey,

I noticed those too, but I do not understand the following code:

event_route [E_UL_CONTACT_REFRESH]
{
 route(PN_SEND, $(param(uri){param.value,pn-provider}),
 $(param(uri){param.value,pn-prid}),
 $(param(uri){param.value,pn-param}));
}

# $param(1) - PN provider
# $param(2) - PN PRID
# $param(3) - PN param
What is this doing? Is the contents of this part lacking? Is it the normal routing code, but the actual pn send is lacking? How do i write exec(what client) or rest_client?

route [PN_SEND]
{
 # send a PN using, for example, **exec** or **rest_client**
}
# how do we actually send the PN?


Regards,

Ronald



January 15, 2024 at 10:57 AM, "Alexey" <slackway2me at gmail.com> wrote:


> 
> Hello,
> 
> there are two tutorials in the official blog:
> 
> part I : https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/
> 
> part II : https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
> 
> -- 
> best regards, Alexey
> https://alexeyka.zantsev.com/
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From rvg at rvgeerligs.nl  Mon Jan 15 16:24:38 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Mon, 15 Jan 2024 16:24:38 +0000
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <6C0F9AFC-196F-40B9-9E30-3ECB53DB65FF@ag-projects.com>
References: <6C0F9AFC-196F-40B9-9E30-3ECB53DB65FF@ag-projects.com>
 <2FE28EAA-BFF2-4913-A19B-BED2789B79FF@ag-projects.com>
 <b238419f5cd995db7c5dec86951d102fb77d87c0@rvgeerligs.nl>
 <53b34a75725ab74eede1afe8b4996afa940729c7@rvgeerligs.nl>
Message-ID: <a2146834331cad69ee3a42f35a55f4136e369b30@rvgeerligs.nl>

Adrian, thank you for the explanation. It is clear to me now. I will firstly go with a solution completely within openSIPS.

Regards,

Ronald



January 15, 2024 at 8:59 AM, "Adrian Georgescu" <ag at ag-projects.com> wrote:


> 
> OpenSIPS handles the SIP traffic, of course. The push server just handles multiple combinations of apps/ push services in an easier way. You could do a lot or all that its needed just from within OpenSIPS but as the number of combinations of apps and services grow another layer of abstraction make it easier to manage, plus you eliminate another possible blocking item from your routes as the works are down asynchronously outside OpenSIPS.
>  
> 
> —
> Adrian
> 
> > 
> > On 14 Jan 2024, at 20:33, rvg at rvgeerligs.nl wrote:
> > 
> > Thank you very much, Adrian, 
> > 
> > I noticed it allready. It seems this is a dedicated push server.
> > Does it handle the normal SIP traffic as well?
> > 
> > Regards,
> > 
> > Ronald
> > 
> > January 14, 2024 at 3:13 PM, "Adrian Georgescu" <ag at ag-projects.com> wrote:
> > 
> > > 
> > > There are many ways of implementing push notifications with OpenSIPS. This is how we implemented it.
> > > 
> > > https://github.com/AGProjects/sylk-pushserver/blob/master/config/opensips.cfg
> > > 
> > > —
> > > Adrian
> > > 
> > > > 
> > > > On 13 Jan 2024, at 17:37, rvg at rvgeerligs.nl wrote:
> > > >  
> > > >  Hi All,
> > > >  
> > > >  
> > > >  I am trying to implement an opensips server 3.4 with pn support. Primarily apple push notifications. Later I will try google.
> > > >  
> > > >  Does any one have this implemented yet?
> > > >  
> > > >  Where can I find an example cfg for opensips?
> > > >  
> > > >  Any help is appreciated.
> > > >  
> > > >  
> > > >  Regards,
> > > >  
> > > >  
> > > >  Ronald Geerligs
> > > >  _______________________________________________
> > > >  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
> >
>
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From rvg at rvgeerligs.nl  Mon Jan 15 18:28:50 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Mon, 15 Jan 2024 18:28:50 +0000
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>
References: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>
Message-ID: <ce636c922bc215c1679fafa9490da41df7047613@rvgeerligs.nl>

hi all,

for your information:
**Sending a push notification to an iOS device using a Bash script**
https://medium.com/@egzon.arifi/sending-a-push-notification-to-an-ios-device-using-a-bash-script-96c056c1544c

I hope the pn in opensips only requires the route to the apn:
https://api.development.push.apple.com/. dev
https://api.push.apple.com/ opp

but I fear it does not.

Regards,

Ronald



January 15, 2024 at 10:57 AM, "Alexey" <slackway2me at gmail.com> wrote:


> 
> Hello,
> 
> there are two tutorials in the official blog:
> 
> part I : https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/
> 
> part II : https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/
> 
> -- 
> best regards, Alexey
> https://alexeyka.zantsev.com/
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From nzdealshelp at gmail.com  Mon Jan 15 19:07:08 2024
From: nzdealshelp at gmail.com (nz deals)
Date: Tue, 16 Jan 2024 08:07:08 +1300
Subject: [OpenSIPS-Users] uac_replace_from in CANCEL
In-Reply-To: <40f20910-463c-4b7d-9bc5-7aa045b02d75@opensips.org>
References: <CALOQYivKCFZYHHxE_TxUyD4dM22BSPA6Y6h_ORzZxh9UK52t_g@mail.gmail.com>
 <7a5cd98b-4123-40d2-8dd2-32b23131d7bd@opensips.org>
 <CALOQYisg-g=6xx=OJ_+1eQHPRE3BFFC9vswYso41wChzLXXZ6g@mail.gmail.com>
 <40f20910-463c-4b7d-9bc5-7aa045b02d75@opensips.org>
Message-ID: <CALOQYitbixpWyr2Wo8-dx_5SbJxd-meyGxWZudjX7cMXdOZQsQ@mail.gmail.com>

Thank you Bogdan,

It seems that replacing remove_hf with uac_replace_from has resolved the
issue.
Just a quick question: if I use remove_hf("Contact") during INVITE, will TM
become aware of this change?



Regards,
Jason

On Fri, 5 Jan 2024 at 23:55, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jason,
>
> For changing the FROM hdr (at INVITE time), better use the
> uac_replace_from() function [1] - by doing this, the TM will become aware
> of the change and reflect it into CANCEL too.
>
>
> [1]
> https://opensips.org/html/docs/modules/3.4.x/uac.html#func_uac_replace_from
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 24.12.2023 13:33, nz deals wrote:
>
> Hi Bogdan,
> Agreed that the RFC3261 CANCEL is truly built using INVITE. In my case,
> somehow CANCEL is not the same as the initial INVITE. Due to the complexity
> of my case I have hard times to fix the issue. Let me explain a bit.
> On the INVITE i change the From Header, which i can see has also been
> changed during Authorization.
> If i CANCEL then the From header is different (it's the same as i have
> before making my change).
>
> I am also using topology_hiding().
>
>
>
> under the main route i change the From header for INVITE
>
>         set_advertised_address("104.13.xx.xx "); # same as i have used in
> the advertised_address
>         # this is my cancel processing
>         if (is_method("CANCEL")) {
>                 if (t_check_trans())
>                         t_relay(8);
>                 exit;
>         }
>         if (has_totag()) {
>             if(topology_hiding_match())  {
>                                    # remove and add a custom one
>                                    remove_hf("From");
>                                    append_hf("From: \"$fU\"
> <sip:$fU@$td;custom=$var(custom)>;tag=$ft\r\n");
>
>  force_send_socket("tls:192.xx.xx.xx:5061");
>              }
>         }
>
> My opensips version is 3.4.2
> My listening socket is socket=tls:ens192:5061 tag ens192     (its the same
> 192.xx.xx.xx IP that you see in the force_send_socket)
> I am also using public ip as follows
> advertised_address=104.13.xx.xx
> alias=104.13.xx.xx
>
>
> Regards,
> Jason
>
> On Thu, 21 Dec 2023 at 02:31, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Jason,
>>
>> In transactional stateful SIP, the CANCEL requests are hop by hop - each
>> hop in the path is generating its own CANCEL requests to the next hop,
>> which consumes it; there is no actual relaying of the CANCELs. So, the
>> replacing (which works in relaying mode only) doesn't fit here.
>>
>> Even more, the RFC3261 gives a rigorous way for building the CANCEL
>> requests, they are to be built 100% based on the INVITE request only
>> (nothing more). OpenSIPS internally builds the CANCEL in accordance to the
>> corresponding INVITE, so you should not need any such changes.
>>
>> May you detail why you think you need this FROM replacement?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>>   https://www.siphub.com
>>
>> On 20.12.2023 12:38, nz deals wrote:
>>
>> Hi,
>> I've been attempting to modify the From header in the CANCEL, but it
>> seems the changes aren't taking effect. I've also experimented with
>> remove_hf and append_hf, but unfortunately, these methods didn't work
>> either. Could someone kindly offer suggestions if I might be overlooking
>> something?
>>
>> This is in the main request route
>>
>>         if (is_method("CANCEL")) {
>>                 uac_replace_from("","sip:test at test.com");
>>                 if (t_check_trans())
>>                         t_relay(8);
>>                 exit;
>>         }
>>
>>
>>
>>
>> Regards,
>> Jason
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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From igorolhovskiy at gmail.com  Mon Jan 15 19:08:27 2024
From: igorolhovskiy at gmail.com (Ihor Olkhovskyi)
Date: Mon, 15 Jan 2024 20:08:27 +0100
Subject: [OpenSIPS-Users] Removing header by contents
In-Reply-To: <CAPc6LUHPmDwOyFQ+fF2wFVCSvnmku9Mzx9QqxdBXoKuz-ADUZg@mail.gmail.com>
References: <CAPc6LUHPmDwOyFQ+fF2wFVCSvnmku9Mzx9QqxdBXoKuz-ADUZg@mail.gmail.com>
Message-ID: <dd4aedee-f6b2-467e-91e2-4f9e8e7a677e@gmail.com>

Thanks,

I've tried to use them, but seems they are taking for both expressions, 
regex and glob not looking in Header values, but names only

When looking at the code, my suspicion getting stronger

https://github.com/OpenSIPS/opensips/blob/224a3236cbeaeb06a897cb3beab8c3a0c3d5903f/modules/sipmsgops/sipmsgops.c#L425

Le 15/01/2024 à 15:03, Alexey a écrit :
> Hello Igor,
>
> these functions - isn't it what you need?
>
> remove_hf_re(hname_expr) [1]
> remove_hf_glob(hname_pattern) [2]
>
> 1. https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_re
> 2. https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_glob
>
>


From rvg at rvgeerligs.nl  Mon Jan 15 21:12:30 2024
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Mon, 15 Jan 2024 21:12:30 +0000
Subject: [OpenSIPS-Users] opensips 3.4 documentation event_routing example
 1.6.1 - push notification
Message-ID: <a1babe16207cd061e6a5c6c603981163e8a36e0b@rvgeerligs.nl>

Hi openSIPS team,

I do not understand the documentaion (opensips version 3.4 manual) on the event_routing module.

The example given 1.6.1 push notification does not seem to do just that.

It says:
"
1.6.1. Push Notification


We use*notify_on_event*to capture the events on new contact registrations for callee. Once the call is sent to callee, based on the notification (for new contacts) we inject the newly registered contacts as new branches in the ongoing transaction.



Schematics : when we send a call to a user, we subscribe to see any new contacts being registered by the user. On such a notification, we add the new contact as a new branch to the ongoing transaction (ringing) to user.

"
It seems this has to do with notify_on_event.
It may be that the new branches are be added with device information for the push notification, but I do not see that in the example given.

Please advise,

Regards,

Ronald Geerligs
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From bogdan at opensips.org  Tue Jan 16 07:35:46 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 16 Jan 2024 09:35:46 +0200
Subject: [OpenSIPS-Users] uac_replace_from in CANCEL
In-Reply-To: <CALOQYitbixpWyr2Wo8-dx_5SbJxd-meyGxWZudjX7cMXdOZQsQ@mail.gmail.com>
References: <CALOQYivKCFZYHHxE_TxUyD4dM22BSPA6Y6h_ORzZxh9UK52t_g@mail.gmail.com>
 <7a5cd98b-4123-40d2-8dd2-32b23131d7bd@opensips.org>
 <CALOQYisg-g=6xx=OJ_+1eQHPRE3BFFC9vswYso41wChzLXXZ6g@mail.gmail.com>
 <40f20910-463c-4b7d-9bc5-7aa045b02d75@opensips.org>
 <CALOQYitbixpWyr2Wo8-dx_5SbJxd-meyGxWZudjX7cMXdOZQsQ@mail.gmail.com>
Message-ID: <f5bf02d6-ec11-404b-b45e-7549652c0899@opensips.org>

The Contact hdr is not relevant for TM, still it is if you are using 
nathelper (for pinging) or dialog support with topo hiding (where the Ct 
must be retained).

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 15.01.2024 21:07, nz deals wrote:
> Thank you Bogdan,
>
> It seems that replacing remove_hf with uac_replace_from has resolved 
> the issue.
> Just a quick question: if I use remove_hf("Contact") during INVITE, 
> will TM become aware of this change?
>
>
>
> Regards,
> Jason
>
> On Fri, 5 Jan 2024 at 23:55, Bogdan-Andrei Iancu <bogdan at opensips.org> 
> wrote:
>
>     Hi Jason,
>
>     For changing the FROM hdr (at INVITE time), better use the
>     uac_replace_from() function [1] - by doing this, the TM will
>     become aware of the change and reflect it into CANCEL too.
>
>
>     [1]
>     https://opensips.org/html/docs/modules/3.4.x/uac.html#func_uac_replace_from
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com
>        https://www.siphub.com
>
>     On 24.12.2023 13:33, nz deals wrote:
>>     Hi Bogdan,
>>     Agreed that the RFC3261 CANCEL is truly built using INVITE. In my
>>     case, somehow CANCEL is not the same as the initial INVITE. Due
>>     to the complexity of my case I have hard times to fix the issue.
>>     Let me explain a bit.
>>     On the INVITE i change the From Header, which i can see has also
>>     been changed during Authorization.
>>     If i CANCEL then the From header is different (it's the same as i
>>     have before making my change).
>>
>>     I am also using topology_hiding().
>>
>>
>>     under the main route i change the From header for INVITE
>>
>>             set_advertised_address("104.13.xx.xx "); # same as i have
>>     used in the advertised_address
>>             # this is my cancel processing
>>             if (is_method("CANCEL")) {
>>                     if (t_check_trans())
>>                             t_relay(8);
>>                     exit;
>>             }
>>             if (has_totag()) {
>>                 if(topology_hiding_match())  {
>>                                        # remove and add a custom one
>>      remove_hf("From");
>>                                        append_hf("From:
>>     \"$fU\"<sip:$fU@$td;custom=$var(custom)>;tag=$ft\r\n");
>>      force_send_socket("tls:192.xx.xx.xx:5061");
>>                  }
>>             }
>>     My opensips version is 3.4.2
>>     My listening socket is socket=tls:ens192:5061 tag ens192     (its
>>     the same 192.xx.xx.xx IP that you see in the force_send_socket)
>>     I am also using public ip as follows
>>     advertised_address=104.13.xx.xx
>>     alias=104.13.xx.xx
>>
>>     Regards,
>>     Jason
>>
>>     On Thu, 21 Dec 2023 at 02:31, Bogdan-Andrei Iancu
>>     <bogdan at opensips.org> wrote:
>>
>>         Hi Jason,
>>
>>         In transactional stateful SIP, the CANCEL requests are hop by
>>         hop - each hop in the path is generating its own CANCEL
>>         requests to the next hop, which consumes it; there is no
>>         actual relaying of the CANCELs. So, the replacing (which
>>         works in relaying mode only) doesn't fit here.
>>
>>         Even more, the RFC3261 gives a rigorous way for building the
>>         CANCEL requests, they are to be built 100% based on the
>>         INVITE request only (nothing more). OpenSIPS internally
>>         builds the CANCEL in accordance to the corresponding INVITE,
>>         so you should not need any such changes.
>>
>>         May you detail why you think you need this FROM replacement?
>>
>>         Regards,
>>
>>         Bogdan-Andrei Iancu
>>
>>         OpenSIPS Founder and Developer
>>            https://www.opensips-solutions.com
>>            https://www.siphub.com
>>
>>         On 20.12.2023 12:38, nz deals wrote:
>>>         Hi,
>>>         I've been attempting to modify the From header in the
>>>         CANCEL, but it seems the changes aren't taking effect. I've
>>>         also experimented with remove_hf and append_hf, but
>>>         unfortunately, these methods didn't work either. Could
>>>         someone kindly offer suggestions if I might be overlooking
>>>         something?
>>>
>>>         This is in the main request route
>>>
>>>                 if (is_method("CANCEL")) {
>>>                         uac_replace_from("","sip:test at test.com
>>>         <mailto:sip%3Atest at test.com>");
>>>                         if (t_check_trans())
>>>                                 t_relay(8);
>>>                         exit;
>>>                 }
>>>
>>>
>>>
>>>         Regards,
>>>         Jason
>>>
>>>         _______________________________________________
>>>         Users mailing list
>>>         Users at lists.opensips.org
>>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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From 342899761 at qq.com  Wed Jan 17 03:05:04 2024
From: 342899761 at qq.com (=?ISO-8859-1?B?bm9uYW1l?=)
Date: Wed, 17 Jan 2024 11:05:04 +0800
Subject: [OpenSIPS-Users] parameters missing with event_datagram.so
Message-ID: <tencent_9F9C38A1049E8102B7349E598893912CEC09@qq.com>

Hi, OpenSIPS experts:


OpenSIPS3.2.16
CentOS7.9


start `nc -u -l 1234`
start `opensips`


```
{"jsonrpc":"2.0","method":"E_MY_EVENT"}
```
parameters missing.


opensips.cfg:


```
log_level=3
xlog_level=3
log_stderror=yes
&nbsp;
socket=udp:127.0.0.1:5060
mpath="/usr/lib64/opensips/modules"
&nbsp;
loadmodule "signaling.so"
loadmodule "sl.so"
&nbsp;
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)
&nbsp;
loadmodule "event_datagram.so"
loadmodule "proto_udp.so"
&nbsp;
route{
	send_reply(200,"OK");
	exit;
}
&nbsp;
startup_route{
	subscribe_event("E_MY_EVENT", "udp:127.0.0.1:1234");
	route(FIRE_TEST);
}
&nbsp;
route[FIRE_TEST]{
	$avp(attr) = 3;
	$avp(attr) = 5;
	raise_event("E_MY_EVENT", $avp(attr));
}

```
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From igorolhovskiy at gmail.com  Wed Jan 17 09:53:59 2024
From: igorolhovskiy at gmail.com (Ihor Olkhovskyi)
Date: Wed, 17 Jan 2024 10:53:59 +0100
Subject: [OpenSIPS-Users] opensips 3.4 documentation event_routing
 example 1.6.1 - push notification
In-Reply-To: <a1babe16207cd061e6a5c6c603981163e8a36e0b@rvgeerligs.nl>
References: <a1babe16207cd061e6a5c6c603981163e8a36e0b@rvgeerligs.nl>
Message-ID: <CAJTkRNsUW6dg11_r2Z7gx9ZoKm54u3AfBXR1SUksF3XddOCB4g@mail.gmail.com>

Ronald,

Have you checked https://www.youtube.com/watch?v=Bx0_l0ONa0g ?

Le lun. 15 janv. 2024 à 22:14, <rvg at rvgeerligs.nl> a écrit :

> Hi openSIPS team,
>
> I do not understand the documentaion (opensips version 3.4 manual) on the
> event_routing module.
>
> The example given 1.6.1 push notification does not seem to do just that.
>
> It says:
> "
> 1.6.1. Push Notification
>
> We use*notify_on_event*to capture the events on new contact registrations
> for callee. Once the call is sent to callee, based on the notification (for
> new contacts) we inject the newly registered contacts as new branches in
> the ongoing transaction.
>
> Schematics : when we send a call to a user, we subscribe to see any new
> contacts being registered by the user. On such a notification, we add the
> new contact as a new branch to the ongoing transaction (ringing) to user.
> "
> It seems this has to do with notify_on_event.
> It may be that the new branches are be added with device information for
> the push notification, but I do not see that in the example given.
>
> Please advise,
>
> Regards,
>
> Ronald Geerligs
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Best regards,
Ihor (Igor)
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From razvan at opensips.org  Thu Jan 18 10:27:37 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Thu, 18 Jan 2024 12:27:37 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
Message-ID: <319ad443-c099-421c-a901-e20db26d9890@opensips.org>

HI, Trevor!

Unfortunately there is currently no way to round up the value, but what 
is unclear is why you need the extra billed_duration, and not simply 
using the duration field.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 1/11/24 15:55, trevor at webon.co.za wrote:
> Hi All,
> 
> I have a simple issue I need to solve but cant seem to fine a elegant
> way of resolving.
> 
> I am using acc module + rate_cacher to do some simple billing.
> 
> I use $DLG_lifetime but this value seems to be a rounded to the nearest
> second what I need is to round up to the nearest second similar to how
> ACC module calculates duration
> 
> Here is an example CDR
> 
> 
> *************************** 1. row ***************************
>         duration: 35
>      ms_duration: 34168
> billed_duration: 34
> 
> As you can see duration is rounded up and billed_duration is rounded
> down.
> 
> Is there any way to get access to the ms of the dialogue from any
> module, I could resort to initializing my own timers but there is
> already a timer running in the acc module and would be more efficient
> to just use it.
> 
> 
> Here is how I am creating billed_duration
> 
> if (has_totag()) {
>      ....
>      ....
>      ....
>      if (is_method("BYE")) {
>          if ($DLG_lifetime == 0) {
>              $acc_extra(billed_duration) = 1;
>          }else{
>              $acc_extra(billed_duration) = $DLG_lifetime;
>          }
>      }
> 
>      route(RELAY);
>      exit;
> }
> 
> 
> Thanks
> Trevor Steyn
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From razvan at opensips.org  Thu Jan 18 10:32:39 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Thu, 18 Jan 2024 12:32:39 +0200
Subject: [OpenSIPS-Users] Removing header by contents
In-Reply-To: <90b1cb29-f628-4480-b789-155897126d0c@gmail.com>
References: <90b1cb29-f628-4480-b789-155897126d0c@gmail.com>
Message-ID: <6d257cab-c798-4372-abe7-6f0590e4968d@opensips.org>

Unfortunately there is no support for this in OpenSIPS - you can only 
remove a header by its name, and that would remove all occurrences.
I guess your only solution is what you already did: remove all and add 
back only what you want.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 1/14/24 19:00, Ihor Olkhovskyi wrote:
> Hello!
> 
> Is there any easy possibility to remove header not by name, but name and 
> contents or just header number?
> 
> I have several Route headers and want to remove the first one only.
> 
> For the moment I'm thinking to remove all headers and add all but the 
> first in the loop, but maybe it's an overkill?
> 
> P.S.: Why I'm doing this - trying to implement own loose_route for some 
> experiments
> 
> Thanks in advance,
> Ihor
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From akogan at 5gfuture.com  Thu Jan 18 10:51:40 2024
From: akogan at 5gfuture.com (Alexander Kogan)
Date: Thu, 18 Jan 2024 14:51:40 +0400
Subject: [OpenSIPS-Users] Error in re.subst transformation
Message-ID: <7d4914e4-0de5-41c8-9a01-d61c61a96d7f@5gfuture.com>

Hi,

There's an error in re.subst transformation, it's not whitespace-safe. 
Is this patch OK?

Index: transformations.c
===================================================================
--- transformations.c   (revision 13886)
+++ transformations.c   (working copy)
@@ -3630,7 +3630,7 @@
                 }
                 p++;
                 LM_DBG("preparing to parse param\n");
-               if (tr_parse_sparam(p, in, &tp, 0) == NULL)
+               if (tr_parse_sparam(p, in, &tp, 1) == NULL)
                         goto error;
                 t->params = tp;
                 tp = 0;

-- 
Best regards,
Alexander Kogan,
Director of R&D
5g Future
http://5gfuture.com



From liviu at opensips.org  Thu Jan 18 12:48:38 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Thu, 18 Jan 2024 14:48:38 +0200
Subject: [OpenSIPS-Users] Error in re.subst transformation
In-Reply-To: <7d4914e4-0de5-41c8-9a01-d61c61a96d7f@5gfuture.com>
References: <7d4914e4-0de5-41c8-9a01-d61c61a96d7f@5gfuture.com>
Message-ID: <47a59614-e9e0-1044-8406-803141a10dad@opensips.org>

On 18.01.2024 12:51, Alexander Kogan wrote:
> Hi,
>
> There's an error in re.subst transformation, it's not whitespace-safe. 
> Is this patch OK?
>
> Index: transformations.c
> ===================================================================
> --- transformations.c   (revision 13886)
> +++ transformations.c   (working copy)
> @@ -3630,7 +3630,7 @@
>                 }
>                 p++;
>                 LM_DBG("preparing to parse param\n");
> -               if (tr_parse_sparam(p, in, &tp, 0) == NULL)
> +               if (tr_parse_sparam(p, in, &tp, 1) == NULL)
>                         goto error;
>                 t->params = tp;
>                 tp = 0;
>
Hi Alexander,

Looks OK to me, for example this quick test seems to PASS after your 
patch without any side-effects on other tests:

     $var(in) = "+443 493 2212";
     assert($(var(in){re.subst, /([0-9]) ([0-9])/\1.\2/g}) == 
"+443.493.2212", "test-subst-2");

If you do a Pull Request on `master`, it will most likely get merged.  
For the stable branches, this limitation can be bypassed by assigning 
the regex to a variable:

     $var(in) = "+443 493 2212";
     $var(re) = "/([0-9]) ([0-9])/\1.\2/g";
     assert($(var(in){re.subst, $var(re)}) == "+443.493.2212", 
"test-subst-2");

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com
OpenSIPS Summit 2024 May 14-17 Valencia | www.opensips.org/events



From iamhalje at gmail.com  Fri Jan 19 18:01:24 2024
From: iamhalje at gmail.com (Dmitry Ponomaryov)
Date: Fri, 19 Jan 2024 23:01:24 +0500
Subject: [OpenSIPS-Users] mid_register and its possible bad actions with
 De-REGISTER
In-Reply-To: <1ca1f2c0-1e80-bd67-a11c-b1bffb95a70b@gmail.com>
References: <716391af-fb1a-01a9-e440-34a6e31ec45d@gmail.com>
 <a1d0f332-8195-4b62-b51c-6939712898ee@opensips.org>
 <1ca1f2c0-1e80-bd67-a11c-b1bffb95a70b@gmail.com>
Message-ID: <8fc0bb96-d372-4c07-a5df-a45d168c9fd9@gmail.com>

Hi OpenSIPS Community, I would like to try to summarize this post as I 
have often been helped by posts in this maillists, maybe mine will help 
someone else too...

Problem:
Was that everything was built on the fact that the backend(main 
registrar) as Asterisk on chan_sip, was accepting registrations via 
mid_registrar with one SIP-number from the same client but from 
different devices, and in this case when the client has a landline and a 
mobile phone for example, mobile phone as it was said in the problem can 
very often re-register with what actually and fights OpenSIPS, when send 
the function callback to usrloc, from which comes the very de-register, 
so this de-register breaks in chan_sip peer, when under one SIP-number 
registered several devices.

Here, of course, we can approach with different workarounds, such as 
transferring registrations to OpenSIPS, switching from chan_sip to 
res_pjsip... and other types of solutions that will lead you to solving 
new problems, but how could you do without it ))

I've looked at quite a few PN question, video/forums/maillists, and in 
total I've managed for Linphone via usrloc/registrar/mid_registrar to 
send push notification, at least for now, on Android, via 
gorush/Firebase, and in total I thought my messages about disabling in 
modparam some callback functions are not important enough, which are 
probably available to OpenSIPS developers, that is, the problem can 
really be tried to solve in another way, well, or make changes in the 
issue at once in the code, which will be checked like code review from 
project developers : )

Cheers.

P.S thanks Johan for a short but revealing answer about PN.
 > or you don't support push notifications.


On 1/9/2024 10:15 AM, Dmitry Ponomaryov wrote:

> Hello Liviu, In my case, AoR throttling was used, before issue (1), do 
> you also think that this is how it should work? (looks like a 
> CRITICAL) > On the other hand, AoR throttling goes beyond this and 
> just makes sure there is always 1 AoR registered downstream. So on a 
> "contact replace" operation, no SIP signaling should reach the 
> Asterisk in that case. the point is that if different devices are 
> registered under the same AoR, one such De-REGISTER, on the final 
> Asterisk completely deletes the peer, although in some of the 
> locations there is still data and if not for the De-REGISTER, Asterisk 
> would have sent its INVITE, we would go through lookup and call all 
> the devices we need from location via some route, this is the problem 
> that we don’t always need De-REGISTER if there is more than one device 
> under the same AoR(when we don't just reduce REGISTER to the main 
> registrar, but use them as REGISTER from diff devices), which breaks 
> peers on Asterisk with De-REGISTER.
>
> but in total, it might be worth making such modparam module 
> mid_registrar to disable De-REGISTER from the OpenSIPS itself, let the 
> registrations themselves expire on the main registrar, and if there is 
> a new registration with a new ctid with ct-throttling schema, then we 
> will simply update peer again... or if someone needs this feature, 
> then let it be possible to choose an interface and not just, for 
> example: socket=1.1.1.1:5060 socket=2.2.2.2:5060 in this case 
> De-REGISTER will go with 1.1.1.1:5060 on main registrar, because it is 
> simply the first one listed socket in cfg. off course, your opinion is 
> needed here, you are developer of this module since 2016 ;) Best regards!
>
> (1): https://github.com/OpenSIPS/opensips/issues/3193
>
> On 1/8/24 14:40, Liviu Chircu wrote:
>> On 27.12.2023 11:38, Dmitry Ponomaryov wrote:
>>>
>>> All this to say, it might make sense to add the ability to disable 
>>> this De-REGISTER, because with a new registration, we will still 
>>> send it a registration with an updated contact before the main 
>>> registrar times out, and there will be no problems, that is, for 
>>> example, the following settings:
>>>
>>> modparam("mid_registrar", "outgoing_expires", 600) 
>>> modparam("mid_registrar", "default_expires", 300) 
>>
>> Hello Dmitry,
>>
>> From your scenario, it sounds like maybe you should enable AoR 
>> throttling, not just Ct throttling? Because in Ct throttling, each 
>> outgoing contact has a unique "ctid=" parameter, so it must be 
>> de-registered from the main registrar on each contact replace, since 
>> the replacement Contact has a new "ctid=", making it an entirely 
>> different SIP URI.
>>
>> On the other hand, AoR throttling goes beyond this and just makes 
>> sure there is always 1 AoR registered downstream.  So on a "contact 
>> replace" operation,  no SIP signaling should reach the Asterisk in 
>> that case.
>>
>> Best regards.


From rob.dyck at telus.net  Sun Jan 21 17:41:54 2024
From: rob.dyck at telus.net (Robert Dyck)
Date: Sun, 21 Jan 2024 09:41:54 -0800
Subject: [OpenSIPS-Users] Question about rls_presentity table
Message-ID: <3306174.44csPzL39Z@leno.mylan>

Currently using opensips-3.4.3. I am experimenting with using resource lists 
for presence. I have a question about table rls_presentity. When a UA 
subscribes to a presentity using an entry in a resource list an entry is 
created in the DB table rls_presentity. When the presentity sends publish its 
state is added to the entry in rls_presentity.

What I find peculiar, there seems to be no way for opensips to delete the 
entry. If both the watcher/subscriber and the presentity are offline the entry 
in the table remains. Only the state is removed.

Is this intended behaviout?




From rob.dyck at telus.net  Sun Jan 21 18:44:52 2024
From: rob.dyck at telus.net (Robert Dyck)
Date: Sun, 21 Jan 2024 10:44:52 -0800
Subject: [OpenSIPS-Users] Question about rls_presentity table
Message-ID: <2282451.iZASKD2KPV@leno.mylan>

I was too hasty. The subscriber has a long expiry on it's subscribe ( 1 hr ) 
and it is not configurable. The entry was eventually deleted.

Currently using opensips-3.4.3. I am experimenting with using resource lists 
for presence. I have a question about table rls_presentity. When a UA 
subscribes to a presentity using an entry in a resource list an entry is 
created in the DB table rls_presentity. When the presentity sends publish its 
state is added to the entry in rls_presentity.

What I find peculiar, there seems to be no way for opensips to delete the 
entry. If both the watcher/subscriber and the presentity are offline the entry 
in the table remains. Only the state is removed.

Is this intended behaviout?




From bogdan at opensips.org  Mon Jan 22 09:25:37 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 22 Jan 2024 11:25:37 +0200
Subject: [OpenSIPS-Users] DNS could not resolve hostname in dispatcher
In-Reply-To: <PAVPR02MB92071B212CD91AABEB4C9684836C2@PAVPR02MB9207.eurprd02.prod.outlook.com>
References: <PAVPR02MB92073CE6871A6D91E6DC52E8836A2@PAVPR02MB9207.eurprd02.prod.outlook.com>
 <50525c12-d3f9-4f3a-98d6-c8478248dd7b@opensips.org>
 <PAVPR02MB92071B212CD91AABEB4C9684836C2@PAVPR02MB9207.eurprd02.prod.outlook.com>
Message-ID: <576fc0af-615f-48a6-b1af-9baaf706cea4@opensips.org>

Thanks, I will check and follow on github!

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 15.01.2024 10:19, guillaume.desgeorge at orange.com wrote:
>
> Hi Bogdan,
>
> Thanks for your answer.
>
> Issue opened : #3285
>
> Regards,
>
> Guillaume Desgeorge
>
> *De :*Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Envoyé :* mercredi 10 janvier 2024 17:28
> *À :* OpenSIPS users mailling list <users at lists.opensips.org>; 
> DESGEORGE Guillaume INNOV/IT-S <guillaume.desgeorge at orange.com>
> *Objet :* Re: [OpenSIPS-Users] DNS could not resolve hostname in 
> dispatcher
>
> Hi Guillaume,
>
> Indeed, a weird one, not marking the destination as failed upon 
> failure to ping it. I checked the drouting module and it is doing such 
> disable, so it might be a overlook in the dispatcher module. Please 
> open a bug report on the github tracker, so we can address this issue.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
>    https://www.siphub.com
>
> On 09.01.2024 11:32, guillaume.desgeorge at orange.com wrote:
>
>     Hi everyone,
>
>     We are facing a problem with Opensips 3.3 (but I’ve done tests on
>     opensips 2.4 and it was the same).
>
>     In the dispatcher, if we have a name instead of an IP address in
>     “destination”, and that the DNS falls down, we have an error
>     “could not resolve hostname” in the log. Then, Opensips can’t
>     resolve the name, there is no more OPTIONS message sent to the
>     destination but the state stays as 0 and if a call arrives,
>     Opensips still route the INVITE to the last known IP address.
>
>     I find that’s strange :
>
>      1. Either Opensips consider that the DNS is down but the
>         destination may be still up with the last IP address he knows
>         and he could still send OPTIONS messages to the last known IP
>         address
>      2. Either Opensips consider that the DNS is down and doesn’t send
>         OPTIONS message but in that case he should not send INVITE to
>         the old IP address.
>
>     I work with a product that has his own DNS and removes the name of
>     the server in the DNS when the machine is down. So everytime the
>     server is down, Opensips can’t resolve the name and still send
>     calls to that server…
>
>     Is there any parameters I can use to change that way of working ?
>
>     Regards,
>
>         Guillaume
>
>     Orange Restricted
>
>     ____________________________________________________________________________________________________________
>
>     Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
>
>     pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
>
>     a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
>
>     Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
>     This message and its attachments may contain confidential or privileged information that may be protected by law;
>
>     they should not be distributed, used or copied without authorisation.
>
>     If you have received this email in error, please notify the sender and delete this message and its attachments.
>
>     As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
>
>     Thank you.
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ____________________________________________________________________________________________________________
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged information that may be protected by law;
> they should not be distributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and delete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have been modified, changed or falsified.
> Thank you.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From bogdan at opensips.org  Mon Jan 22 11:17:27 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 22 Jan 2024 13:17:27 +0200
Subject: [OpenSIPS-Users] Removing header by contents
In-Reply-To: <dd4aedee-f6b2-467e-91e2-4f9e8e7a677e@gmail.com>
References: <CAPc6LUHPmDwOyFQ+fF2wFVCSvnmku9Mzx9QqxdBXoKuz-ADUZg@mail.gmail.com>
 <dd4aedee-f6b2-467e-91e2-4f9e8e7a677e@gmail.com>
Message-ID: <a083ce9e-cf9e-441c-9c24-eb4f60d132dc@opensips.org>

Indeed, as per docs, the RE is applied only to the hdr name, not to its 
body.

Of course, you can iterate thru the Contact hdrs using the 
$(hdr(Contact)[idx]) and check its value in script, but afterall there 
is no mechanism to delete that particular instance of CT hdr....

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 15.01.2024 21:08, Ihor Olkhovskyi wrote:
> Thanks,
>
> I've tried to use them, but seems they are taking for both 
> expressions, regex and glob not looking in Header values, but names only
>
> When looking at the code, my suspicion getting stronger
>
> https://github.com/OpenSIPS/opensips/blob/224a3236cbeaeb06a897cb3beab8c3a0c3d5903f/modules/sipmsgops/sipmsgops.c#L425 
>
>
> Le 15/01/2024 à 15:03, Alexey a écrit :
>> Hello Igor,
>>
>> these functions - isn't it what you need?
>>
>> remove_hf_re(hname_expr) [1]
>> remove_hf_glob(hname_pattern) [2]
>>
>> 1. 
>> https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_re
>> 2. 
>> https://opensips.org/docs/modules/3.4.x/sipmsgops.html#func_remove_hf_glob
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Mon Jan 22 11:21:43 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 22 Jan 2024 13:21:43 +0200
Subject: [OpenSIPS-Users] parameters missing with event_datagram.so
In-Reply-To: <tencent_9F9C38A1049E8102B7349E598893912CEC09@qq.com>
References: <tencent_9F9C38A1049E8102B7349E598893912CEC09@qq.com>
Message-ID: <e8472d1e-2a8b-450a-884a-a0d79ea4ee58@opensips.org>

Hi,

Please carefully read the doc:
https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#raise_event

/vals/ (var, optional) - AVP containing values attached to the event; if 
this parameter is missing, the raised event will not have any 
attributes, even if the /attrs/ parameter is provided.

^^^ this is your case, passing only "attrs", no "vals"

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 17.01.2024 05:05, noname via Users wrote:
> Hi, OpenSIPS experts:
>
> OpenSIPS3.2.16
> CentOS7.9
>
> start `nc -u -l 1234`
> start `opensips`
>
> ```
> {"jsonrpc":"2.0","method":"E_MY_EVENT"}
> ```
> parameters missing.
>
> opensips.cfg:
>
> ```
> log_level=3
> xlog_level=3
> log_stderror=yes
> socket=udp:127.0.0.1:5060
> mpath="/usr/lib64/opensips/modules"
> loadmodule "signaling.so"
> loadmodule "sl.so"
> loadmodule "tm.so"
> modparam("tm", "fr_timeout", 5)
> modparam("tm", "fr_inv_timeout", 30)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
> loadmodule "event_datagram.so"
> loadmodule "proto_udp.so"
> route{
> send_reply(200,"OK");
> exit;
> }
> startup_route{
> subscribe_event("E_MY_EVENT", "udp:127.0.0.1:1234");
> route(FIRE_TEST);
> }
> route[FIRE_TEST]{
> $avp(attr) = 3;
> $avp(attr) = 5;
> raise_event("E_MY_EVENT", $avp(attr));
> }
> ```
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From bogdan at opensips.org  Mon Jan 22 11:28:43 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 22 Jan 2024 13:28:43 +0200
Subject: [OpenSIPS-Users] opensips 3.4 documentation event_routing
 example 1.6.1 - push notification
In-Reply-To: <a1babe16207cd061e6a5c6c603981163e8a36e0b@rvgeerligs.nl>
References: <a1babe16207cd061e6a5c6c603981163e8a36e0b@rvgeerligs.nl>
Message-ID: <cea68010-f8b7-4f83-b310-4f623e13a050@opensips.org>

Hi Ronald,

Thanks to the "notify_on_event", whenever there is a new registration 
(see E_UL_CONTACT_INSERT) for that user (see the $avp(filter)), the 
"fork_call" route will be execute and a new branch (for that call) will 
be create toward the new registration - see the t_inject_branches() docs.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 15.01.2024 23:12, rvg at rvgeerligs.nl wrote:
> Hi openSIPS team,
>
> I do not understand the documentaion (opensips version 3.4 manual) on 
> the event_routing module.
>
> The example given 1.6.1 push notification does not seem to do just that.
>
> It says:
> "
> 1.6.1. |Push Notification|
>
> We use/notify_on_event/to capture the events on new contact 
> registrations for callee. Once the call is sent to callee, based on 
> the notification (for new contacts) we inject the newly registered 
> contacts as new branches in the ongoing transaction.
>
> Schematics : when we send a call to a user, we subscribe to see any 
> new contacts being registered by the user. On such a notification, we 
> add the new contact as a new branch to the ongoing transaction 
> (ringing) to user.
>
> "
> It seems this has to do with notify_on_event.
> It may be that the new branches are be added with device information 
> for the push notification, but I do not see that in the example given.
>
> Please advise,
>
> Regards,
>
> Ronald Geerligs
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From liviu at opensips.org  Mon Jan 22 12:29:50 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 22 Jan 2024 14:29:50 +0200
Subject: [OpenSIPS-Users] pn push notifications
In-Reply-To: <ce636c922bc215c1679fafa9490da41df7047613@rvgeerligs.nl>
References: <CAPc6LUHvASXR=mKvGSAuC3_dXwF718OhRrsuE7sxbR=6PLOkXg@mail.gmail.com>
 <ce636c922bc215c1679fafa9490da41df7047613@rvgeerligs.nl>
Message-ID: <e3c7ae92-b1f4-488c-9925-b60fcc675e51@opensips.org>

On 15.01.2024 20:28, rvg at rvgeerligs.nl wrote:
> for your information:
> *Sending a push notification to an iOS device using a Bash script*
> https://medium.com/@egzon.arifi/sending-a-push-notification-to-an-ios-device-using-a-bash-script-96c056c1544c
>
> I hope the pn in opensips only requires the route to the apn:
> https://api.development.push.apple.com/. dev
> https://api.push.apple.com/ opp

Hello Ronald,

Indeed, the /PN_SEND /route will most likely invoke a shell script 
(Bash, Go, Python, etc.) which triggers the notification for that 
/specific /SIP device via APNS, using your app developer API 
credentials.  As this logic is not only specific to the mobile 
developer's platform (APNS or Firebase?), but it's also subject to 
change and often depends on the OS as well (Android 11 or 14 API?  iOS 
14 or 16? etc.), we decided to not officially maintain such scripts and 
leave this part to the VoIP platform developer.

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com
OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events
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From razvan at opensips.org  Mon Jan 22 15:05:55 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Mon, 22 Jan 2024 17:05:55 +0200
Subject: [OpenSIPS-Users] OpenSIPS IMS at Fosdem'24
Message-ID: <e830ea1d-2e1e-449f-8933-a4d93463f65d@opensips.org>

Hi, Everyone!

This year history repeats itself, thus myself (Răzvan Crainea) and Liviu 
Chircu will be representing OpenSIPS at the Fosdem'24 conference, where 
will be talking about how you can Provide VoLTE and/or VoNR for IMS 
using OpenSIPS 3.5.
This talk will present the way OpenSIPS is tackling its IMS 
implementation, and will contain a lot of the topics that we developed
within the OpenSIPS IMS Working Group[1] - so if you are interested in 
the topic, make sure you subscribe (if you haven't already) to the group 
and bring your valuable contribution the the IMS topic.
Our presentation starts on Saturday, 17:50[1] in room H.1302, Real Time 
Communications (RTC) devroom[2]. We hope to see as many of you as possible!

[1] http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims
[2] 
https://fosdem.org/2024/schedule/event/fosdem-2024-3614-provide-volte-vonr-using-opensips-3-5/

Happy hacking,
-- 
Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com


From gmaruzz at gmail.com  Mon Jan 22 15:14:36 2024
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 22 Jan 2024 16:14:36 +0100
Subject: [OpenSIPS-Users] [WG-IMS] OpenSIPS IMS at Fosdem'24
In-Reply-To: <e830ea1d-2e1e-449f-8933-a4d93463f65d@opensips.org>
References: <e830ea1d-2e1e-449f-8933-a4d93463f65d@opensips.org>
Message-ID: <CALXCt0qfPhD-9ST57JvB4vv3YBoZssFoCHttWADBw4f07rVzOg@mail.gmail.com>

yeeeeeeeeeeeeee!!!!!!

-giovanni

On Mon, Jan 22, 2024 at 4:06 PM Răzvan Crainea <razvan at opensips.org> wrote:

> Hi, Everyone!
>
> This year history repeats itself, thus myself (Răzvan Crainea) and Liviu
> Chircu will be representing OpenSIPS at the Fosdem'24 conference, where
> will be talking about how you can Provide VoLTE and/or VoNR for IMS
> using OpenSIPS 3.5.
> This talk will present the way OpenSIPS is tackling its IMS
> implementation, and will contain a lot of the topics that we developed
> within the OpenSIPS IMS Working Group[1] - so if you are interested in
> the topic, make sure you subscribe (if you haven't already) to the group
> and bring your valuable contribution the the IMS topic.
> Our presentation starts on Saturday, 17:50[1] in room H.1302, Real Time
> Communications (RTC) devroom[2]. We hope to see as many of you as possible!
>
> [1] http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims
> [2]
>
> https://fosdem.org/2024/schedule/event/fosdem-2024-3614-provide-volte-vonr-using-opensips-3-5/
>
> Happy hacking,
> --
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://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
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From trevor at webon.co.za  Tue Jan 23 07:12:06 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 23 Jan 2024 09:12:06 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
Message-ID: <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>

On Thu, 2024-01-18 at 12:27 +0200, Răzvan Crainea wrote:
> HI, Trevor!
> 
> Unfortunately there is currently no way to round up the value, but
> what 
> is unclear is why you need the extra billed_duration, and not simply 
> using the duration field.
> 
> Best regards,
> 
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
> 
> On 1/11/24 15:55, trevor at webon.co.za wrote:
> > 

Hi Răzvan

I am using rate cacher and want to calculate billing in script and am
unable to use duration field as this is only inserted into DB and not
available as a variable unless I missed something in the document :)

Regards
Trevor Steyn


From bogdan at opensips.org  Tue Jan 23 10:35:03 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 23 Jan 2024 12:35:03 +0200
Subject: [OpenSIPS-Users] regular expression matching question
In-Reply-To: <CA++nn_xKt7rW5CakGw1LBq5s0iurshZysg1FAjHayJ0JhE8=_A@mail.gmail.com>
References: <CA++nn_yyAC1h2vGzaV-L0OTgKywVGb5RC8P7Lob83wUnuXp5fA@mail.gmail.com>
 <CAPWMwMENs-oG7XWxJHMS+Fe16C644CJ5bRivi1=0MDr_j2FJ7w@mail.gmail.com>
 <CA++nn_xKt7rW5CakGw1LBq5s0iurshZysg1FAjHayJ0JhE8=_A@mail.gmail.com>
Message-ID: <d27f80f5-6482-4ee7-b9f8-b97a737bd459@opensips.org>

Hi,

check this https://opensips.org/html/docs/modules/3.4.x/regex.html

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 05.01.2024 13:56, Faheem Muhammad wrote:
> Hi Machel,
>
> Appreciate the correction; it was a typo when I posted the question. 
> However, as Callum mentioned earlier, the "IF" condition with the "=~" 
> operator doesn't fully support PCRE. It seems that any regular 
> expression with parentheses isn't matched using the "=~" operator.
> Using "pcre_match" or an alternative regular expression would be more 
> suitable in this case.
>
> Thank you
> M. Faheem
>
> On Wed, Jan 3, 2024 at 6:37 PM Michel crans <mcrans at gmail.com> wrote:
>
>
>     The reason why your second regular expression is not matching the
>     pattern is because of a typo in the syntax. You have used the
>     |~=| operator instead of the |=~| operator, which is the correct
>     way to test if a string matches a regular expression in PCRE.
>
>
>     Op di 2 jan 2024 om 12:28 schreef Faheem Muhammad
>     <faheem2084 at gmail.com>:
>
>         Hi everyone,
>
>         Firstly, I wish you all a Happy New Year with the best wishes
>         for each community member and the OpenSIPS project.
>
>
>         I have a question related to regular expressions matching in
>         an IF condition.
>         There are two identical regular expressions. One RE is
>         matching and one RE is not matching the pattern.
>
>         *Sample $fU Range: *+966123456700 to +96612345650
>
>         # RE1 - Matching Regular expression
>         if($fU =~"\+9661234567[0-4][0-9]|\+966123456750"){
>         xlog("re matched\n")
>         }
>
>         # RE2  - Non matching Regular expression
>         if($fU ~="\+9661234567([0-4]\d|50)"){
>         } # RE not matched
>
>         Can anyone please suggest what I'm doing wrong?
>
>         *OpenSIPS version*:     3.2.3
>         OS: *Debian 12*
>
>
>         Best regards,
>         M. Faheem
>
>
>         _______________________________________________
>         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
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From marcin at voipplus.net  Tue Jan 23 11:44:07 2024
From: marcin at voipplus.net (Marcin Groszek)
Date: Tue, 23 Jan 2024 05:44:07 -0600
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
Message-ID: <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>

$avp(dlg_lifetime) is a read/ write variable and you can change its 
value before acc is written.

To round up the duration using minimum billing block and increments you 
need to save them first in to a dialog.

this is the logic I use (opensips 3.1)

                                         if ($(DLG_lifetime{s.int}) > 
$(dlg_val(minimum){s.int})) {
math_eval("$DLG_lifetime / $dlg_val(increment)", $avp(lifetime));
math_ceil("$avp(lifetime)", $avp(dlg_total_increment));
math_eval("$avp(dlg_total_increment) * $dlg_val(increment)", 
$avp(dlg_lifetime));
math_ceil("$avp(dlg_lifetime)",$avp(dlg_lifetime));
                                         }
                                         else {
                                                 $avp(dlg_lifetime) = 
$dlg_val(minimum);
                                         }


On 1/23/2024 1:12 AM, trevor at webon.co.za wrote:
> On Thu, 2024-01-18 at 12:27 +0200, Răzvan Crainea wrote:
>> HI, Trevor!
>>
>> Unfortunately there is currently no way to round up the value, but
>> what
>> is unclear is why you need the extra billed_duration, and not simply
>> using the duration field.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Core Developer / SIPhub CTO
>> http://www.opensips-solutions.com / https://www.siphub.com
>>
>> On 1/11/24 15:55, trevor at webon.co.za wrote:
> Hi Răzvan
>
> I am using rate cacher and want to calculate billing in script and am
> unable to use duration field as this is only inserted into DB and not
> available as a variable unless I missed something in the document :)
>
> Regards
> Trevor Steyn
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net



From trevor at webon.co.za  Tue Jan 23 14:29:17 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 23 Jan 2024 16:29:17 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
Message-ID: <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>

On Tue, 2024-01-23 at 05:44 -0600, Marcin Groszek wrote:
> $avp(dlg_lifetime) is a read/ write variable and you can change its 
> value before acc is written.
> 
> To round up the duration using minimum billing block and increments
> you 
> need to save them first in to a dialog.
> 
> this is the logic I use (opensips 3.1)
> 
>                                          if ($(DLG_lifetime{s.int}) >
> $(dlg_val(minimum){s.int})) {
> math_eval("$DLG_lifetime / $dlg_val(increment)", $avp(lifetime));
> math_ceil("$avp(lifetime)", $avp(dlg_total_increment));
> math_eval("$avp(dlg_total_increment) * $dlg_val(increment)", 
> $avp(dlg_lifetime));
> math_ceil("$avp(dlg_lifetime)",$avp(dlg_lifetime));
>                                          }
>                                          else {
>                                                  $avp(dlg_lifetime) =
> $dlg_val(minimum);
>                                          }
> 
> 

Hi Marcin, 

This only works for billing increments bigger than "1" I am
implementing per second billing and $DLG_lifetime is a rounded to the
nearest second, if a call duration is at 1001ms $DLG_Lifetime will
return 1 I need to round up I need to bill anything past 1000ms to 2
seconds.

Regards
Trevor Steyn



From marcin at voipplus.net  Tue Jan 23 15:01:22 2024
From: marcin at voipplus.net (Marcin Groszek)
Date: Tue, 23 Jan 2024 09:01:22 -0600
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
Message-ID: <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>

use math_ceil

1.4.5.  math_ceil(number, result_var)

Truncates a number, always towards +infinity. This means that ceil(3.2) 
= 4.0 and ceil(-2.9) = -2.0


1.10.3. $DLG_lifetime

Returns the duration (in seconds)...



On 1/23/2024 8:29 AM, trevor at webon.co.za wrote:
> On Tue, 2024-01-23 at 05:44 -0600, Marcin Groszek wrote:
>> $avp(dlg_lifetime) is a read/ write variable and you can change its
>> value before acc is written.
>>
>> To round up the duration using minimum billing block and increments
>> you
>> need to save them first in to a dialog.
>>
>> this is the logic I use (opensips 3.1)
>>
>>                                           if ($(DLG_lifetime{s.int}) >
>> $(dlg_val(minimum){s.int})) {
>> math_eval("$DLG_lifetime / $dlg_val(increment)", $avp(lifetime));
>> math_ceil("$avp(lifetime)", $avp(dlg_total_increment));
>> math_eval("$avp(dlg_total_increment) * $dlg_val(increment)",
>> $avp(dlg_lifetime));
>> math_ceil("$avp(dlg_lifetime)",$avp(dlg_lifetime));
>>                                           }
>>                                           else {
>>                                                   $avp(dlg_lifetime) =
>> $dlg_val(minimum);
>>                                           }
>>
>>
> Hi Marcin,
>
> This only works for billing increments bigger than "1" I am
> implementing per second billing and $DLG_lifetime is a rounded to the
> nearest second, if a call duration is at 1001ms $DLG_Lifetime will
> return 1 I need to round up I need to bill anything past 1000ms to 2
> seconds.
>
> Regards
> Trevor Steyn
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net



From marcin at voipplus.net  Tue Jan 23 15:25:27 2024
From: marcin at voipplus.net (Marcin Groszek)
Date: Tue, 23 Jan 2024 09:25:27 -0600
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
Message-ID: <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>

If you are not calculating the cost of the call in the script it self 
you may create external script to go over your acc table and use 
"ms_duration" to adjust "duration"

or use *$Ts**$Tsm *at the beginning of the call, save it to a dialog and 
compare it with same value at the end of the call, then round up and 
write to acc table and/or calculate cost of the call.


**

On 1/23/2024 9:01 AM, Marcin Groszek wrote:
> use math_ceil
>
> 1.4.5.  math_ceil(number, result_var)
>
> Truncates a number, always towards +infinity. This means that 
> ceil(3.2) = 4.0 and ceil(-2.9) = -2.0
>
>
> 1.10.3. $DLG_lifetime
>
> Returns the duration (in seconds)...
>
>
>
> On 1/23/2024 8:29 AM, trevor at webon.co.za wrote:
>> On Tue, 2024-01-23 at 05:44 -0600, Marcin Groszek wrote:
>>> $avp(dlg_lifetime) is a read/ write variable and you can change its
>>> value before acc is written.
>>>
>>> To round up the duration using minimum billing block and increments
>>> you
>>> need to save them first in to a dialog.
>>>
>>> this is the logic I use (opensips 3.1)
>>>
>>>                                           if ($(DLG_lifetime{s.int}) >
>>> $(dlg_val(minimum){s.int})) {
>>> math_eval("$DLG_lifetime / $dlg_val(increment)", $avp(lifetime));
>>> math_ceil("$avp(lifetime)", $avp(dlg_total_increment));
>>> math_eval("$avp(dlg_total_increment) * $dlg_val(increment)",
>>> $avp(dlg_lifetime));
>>> math_ceil("$avp(dlg_lifetime)",$avp(dlg_lifetime));
>>>                                           }
>>>                                           else {
>>> $avp(dlg_lifetime) =
>>> $dlg_val(minimum);
>>>                                           }
>>>
>>>
>> Hi Marcin,
>>
>> This only works for billing increments bigger than "1" I am
>> implementing per second billing and $DLG_lifetime is a rounded to the
>> nearest second, if a call duration is at 1001ms $DLG_Lifetime will
>> return 1 I need to round up I need to bill anything past 1000ms to 2
>> seconds.
>>
>> Regards
>> Trevor Steyn
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net

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From trevor at webon.co.za  Tue Jan 23 19:41:10 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 23 Jan 2024 21:41:10 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
 <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
Message-ID: <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>

On Tue, 2024-01-23 at 09:25 -0600, Marcin Groszek wrote:
>  
> If you are not calculating the cost of the call in the script it self
> you may create external script to go over your acc table and use
> "ms_duration" to adjust "duration"
>  
> or use $Ts$Tsm  at the beginning of the call, save it to a dialog and
> compare it with same value at the end of the call, then round up and
> write to acc table and/or calculate cost of the call.
>  
> 
>  
>  
> On 1/23/2024 9:01 AM, Marcin Groszek wrote:
>  
>  
> > use math_ceil 
> >  
> >  1.4.5.  math_ceil(number, result_var) 
> >  
> >  Truncates a number, always towards +infinity. This means that
> > ceil(3.2) = 4.0 and ceil(-2.9) = -2.0 
> >  
> >  
> >  1.10.3. $DLG_lifetime 
> >  
> >  Returns the duration (in seconds)... 

Hi Marcin, 

Using math_ceil on $DLG_lifetime wont work as it returns seconds not
ms.

As for your other suggestion I am doing something similar just not sure
its the most efficient I am using get_accurate_time would $ts$Tsm be a
better approach,


at 200OK in onreply route 

    if ($rs == 200) {
        get_accurate_time($var(now_s), $var(now_us));
        $dlg_val(start_s) = $var(now_s);
        $dlg_val(start_us) = $var(now_us);
    }

then on BYE

if (is_method("BYE")) {
    get_accurate_time($var(end_s), $var(end_us));
    ts_usec_delta($(dlg_val(start_s){s.int}),
$(dlg_val(start_us){s.int}), $var(end_s),  $var(end_us) ,
$var(billed_duration));
   $var(end_us) , $acc_extra(billed_duration)\n");
    math_eval("$var(billed_duration) / 1000000",
$var(billed_duration));  
    math_ceil($var(billed_duration),$var(billed_duration));
    $acc_extra(billed_duration) = $var(billed_duration);

## Rate the call

    math_eval("$acc_extra(c_price) / 60 * $acc_extra(billed_duration)",
$var(c_cost));
    math_eval("$acc_extra(v_price) / 60 * $acc_extra(billed_duration)",
$var(v_cost));
    $acc_extra(cost) = $var(c_cost);
    $acc_extra(v_cost) = $var(v_cost);
}

Thanks.


From Ben.Newlin at genesys.com  Tue Jan 23 19:59:57 2024
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Tue, 23 Jan 2024 19:59:57 +0000
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
 <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
 <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
Message-ID: <BN0PR10MB5208E2BC0F872C730E5DCB3795742@BN0PR10MB5208.namprd10.prod.outlook.com>

A few people have mentioned this but I haven’t seen a response. Why do you have to do this conversion in the script? The call duration in ms is already included by default in every CDR. We are only talking about rounding here, why wouldn’t your rating application do that for you? That’s much less complex than adding an entirely new field just so OpenSIPS can do the rounding.

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of trevor at webon.co.za <trevor at webon.co.za>
Date: Tuesday, January 23, 2024 at 2:42 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] $DLG_lifetime but in ms
 EXTERNAL EMAIL - Please use caution with links and attachments

On Tue, 2024-01-23 at 09:25 -0600, Marcin Groszek wrote:
>
> If you are not calculating the cost of the call in the script it self
> you may create external script to go over your acc table and use
> "ms_duration" to adjust "duration"
>
> or use $Ts$Tsm  at the beginning of the call, save it to a dialog and
> compare it with same value at the end of the call, then round up and
> write to acc table and/or calculate cost of the call.
>
>
>
>
> On 1/23/2024 9:01 AM, Marcin Groszek wrote:
>
>
> > use math_ceil
> >
> >  1.4.5.  math_ceil(number, result_var)
> >
> >  Truncates a number, always towards +infinity. This means that
> > ceil(3.2) = 4.0 and ceil(-2.9) = -2.0
> >
> >
> >  1.10.3. $DLG_lifetime
> >
> >  Returns the duration (in seconds)...

Hi Marcin,

Using math_ceil on $DLG_lifetime wont work as it returns seconds not
ms.

As for your other suggestion I am doing something similar just not sure
its the most efficient I am using get_accurate_time would $ts$Tsm be a
better approach,


at 200OK in onreply route

    if ($rs == 200) {
        get_accurate_time($var(now_s), $var(now_us));
        $dlg_val(start_s) = $var(now_s);
        $dlg_val(start_us) = $var(now_us);
    }

then on BYE

if (is_method("BYE")) {
    get_accurate_time($var(end_s), $var(end_us));
    ts_usec_delta($(dlg_val(start_s){s.int}),
$(dlg_val(start_us){s.int}), $var(end_s),  $var(end_us) ,
$var(billed_duration));
   $var(end_us) , $acc_extra(billed_duration)\n");
    math_eval("$var(billed_duration) / 1000000",
$var(billed_duration));
    math_ceil($var(billed_duration),$var(billed_duration));
    $acc_extra(billed_duration) = $var(billed_duration);

## Rate the call

    math_eval("$acc_extra(c_price) / 60 * $acc_extra(billed_duration)",
$var(c_cost));
    math_eval("$acc_extra(v_price) / 60 * $acc_extra(billed_duration)",
$var(v_cost));
    $acc_extra(cost) = $var(c_cost);
    $acc_extra(v_cost) = $var(v_cost);
}

Thanks.

_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From marcin at voipplus.net  Tue Jan 23 20:09:13 2024
From: marcin at voipplus.net (Marcin Groszek)
Date: Tue, 23 Jan 2024 14:09:13 -0600
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
 <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
 <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
Message-ID: <8bbe9847-b153-90d3-c165-95a1a2526f0a@voipplus.net>

I am not sure what would be less resource hungry;  $Ts$Tsm v 
get_accurate_time.

get_accurate_time is a function v 2 core variables.
I guess a config can be created to run 100 million of both operation and measuring resources used, cpu, ram and time to complete.

This only makes sense only if you are calculating cost of the each call in the script.
If not then just use external script to round/modifyduration based on ms_duration in acc table after the fact.

On 1/23/2024 1:41 PM, trevor at webon.co.za wrote:
> On Tue, 2024-01-23 at 09:25 -0600, Marcin Groszek wrote:
>>   
>> If you are not calculating the cost of the call in the script it self
>> you may create external script to go over your acc table and use
>> "ms_duration" to adjust "duration"
>>   
>> or use $Ts$Tsm  at the beginning of the call, save it to a dialog and
>> compare it with same value at the end of the call, then round up and
>> write to acc table and/or calculate cost of the call.
>>   
>>
>>   
>>   
>> On 1/23/2024 9:01 AM, Marcin Groszek wrote:
>>   
>>   
>>> use math_ceil
>>>   
>>>   1.4.5.  math_ceil(number, result_var)
>>>   
>>>   Truncates a number, always towards +infinity. This means that
>>> ceil(3.2) = 4.0 and ceil(-2.9) = -2.0
>>>   
>>>   
>>>   1.10.3. $DLG_lifetime
>>>   
>>>   Returns the duration (in seconds)...
> Hi Marcin,
>
> Using math_ceil on $DLG_lifetime wont work as it returns seconds not
> ms.
>
> As for your other suggestion I am doing something similar just not sure
> its the most efficient I am using get_accurate_time would $ts$Tsm be a
> better approach,
>
>
> at 200OK in onreply route
>
>      if ($rs == 200) {
>          get_accurate_time($var(now_s), $var(now_us));
>          $dlg_val(start_s) = $var(now_s);
>          $dlg_val(start_us) = $var(now_us);
>      }
>
> then on BYE
>
> if (is_method("BYE")) {
>      get_accurate_time($var(end_s), $var(end_us));
>      ts_usec_delta($(dlg_val(start_s){s.int}),
> $(dlg_val(start_us){s.int}), $var(end_s),  $var(end_us) ,
> $var(billed_duration));
>     $var(end_us) , $acc_extra(billed_duration)\n");
>      math_eval("$var(billed_duration) / 1000000",
> $var(billed_duration));
>      math_ceil($var(billed_duration),$var(billed_duration));
>      $acc_extra(billed_duration) = $var(billed_duration);
>
> ## Rate the call
>
>      math_eval("$acc_extra(c_price) / 60 * $acc_extra(billed_duration)",
> $var(c_cost));
>      math_eval("$acc_extra(v_price) / 60 * $acc_extra(billed_duration)",
> $var(v_cost));
>      $acc_extra(cost) = $var(c_cost);
>      $acc_extra(v_cost) = $var(v_cost);
> }
>
> Thanks.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net



From trevor at webon.co.za  Tue Jan 23 21:01:38 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 23 Jan 2024 23:01:38 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <BN0PR10MB5208E2BC0F872C730E5DCB3795742@BN0PR10MB5208.namprd10.prod.outlook.com>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
 <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
 <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
 <BN0PR10MB5208E2BC0F872C730E5DCB3795742@BN0PR10MB5208.namprd10.prod.outlook.com>
Message-ID: <fbafdd1525acae88a86ebdcd103a24a5929698d1.camel@webon.co.za>

On Tue, 2024-01-23 at 19:59 +0000, Ben Newlin wrote:
> 
> 
> A few people have mentioned this but I haven’t seen a response. Why
> do you have to do this conversion in the script? The call duration in
> ms is already included by default in every CDR. We are only talking
> about rounding here, why wouldn’t your rating application do that for
> you? That’s much less complex than adding an entirely new field just
> so OpenSIPS can do the rounding.
>  
> 
> Ben Newlin
>  

Hi Benlin,

Yes if you are rating out of opensips the ms_duration field and
duration is already rounded for my liking, But I am not wanting to
build an external billing engine, I want to use Rate cacher (meant to
do simple billing inside opensips) thus need access to accurate time of
dialogue.

Regards
Trevor Steyn


From trevor at webon.co.za  Tue Jan 23 21:03:52 2024
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 23 Jan 2024 23:03:52 +0200
Subject: [OpenSIPS-Users] $DLG_lifetime but in ms
In-Reply-To: <8bbe9847-b153-90d3-c165-95a1a2526f0a@voipplus.net>
References: <44c6cdf3c71f1bf340508ee940a81938f976ed9b.camel@webon.co.za>
 <319ad443-c099-421c-a901-e20db26d9890@opensips.org>
 <bf4a24458a7fbdbf92b52205f7fefea568feef94.camel@webon.co.za>
 <2ad5a9a0-4104-e4a1-8168-8e3269632f54@voipplus.net>
 <bd28a91d2d5426da75c5d9ec29eebabf4f5118b6.camel@webon.co.za>
 <67bce38d-58ea-1ab0-08c9-5ab02ca009f4@voipplus.net>
 <95c37c0a-7fa8-c57b-fd34-9edd618ff6ed@voipplus.net>
 <0c4653c43cdbdebfff348673503cdd872162d5e4.camel@webon.co.za>
 <8bbe9847-b153-90d3-c165-95a1a2526f0a@voipplus.net>
Message-ID: <cdb0567cc9f9c9553341ebf455fe347d5d49eccc.camel@webon.co.za>

On Tue, 2024-01-23 at 14:09 -0600, Marcin Groszek wrote:
> I am not sure what would be less resource hungry;  $Ts$Tsm v 
> get_accurate_time.
> 
> get_accurate_time is a function v 2 core variables.
> I guess a config can be created to run 100 million of both operation
> and measuring resources used, cpu, ram and time to complete.
> 
> This only makes sense only if you are calculating cost of the each
> call in the script.
> If not then just use external script to round/modifyduration based on
> ms_duration in acc table after the fact.
> 
Thanks Marcin, 

yes I think I may benchmark the 2 to see which is more efficeint, yes I
am trying to bill inside of opensips using ratecacher module for this
purpose.

Regards
Trevor Steyn


From spanda at 3clogic.com  Wed Jan 24 10:04:12 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 24 Jan 2024 15:34:12 +0530
Subject: [OpenSIPS-Users] Need some help around script transfermation in
	opensips 3.2 .
Message-ID: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>

Hi All ,
I am using opensips 3.2 script transformation parameters to find the
existence of a  parameter in  ; separated list of parameters .

In INVITE I have a custom header X-tag which contains a value as below .
X-Tag : Laptop

Agents used to get registered with X-Tag with some values and I used to put
them in the attr column of the location table .
Agent X-Tag : Laptop=1;Laptop laptop=1

While processing the call , I am matching if the X-Tag in Invite exists in
the Agent tag . If the exact parameter exists  then give a call to that
contact else not . For that I using below transformation

*{param.exist,Laptop}*


Somehow this is not working . It's giving an error . As for my expectation
it should match with the first parameter and as that's an exact match agent
must get the call . But its not happening and breaking with error .

*ERROR:core:parse_params: invalid character, ; expected, found l*

*while checking the existence of parameters, is that only checking a single
word without any space ?Also special characters are allowed in between or
not  ? *

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From bogdan at opensips.org  Wed Jan 24 12:42:37 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Jan 2024 14:42:37 +0200
Subject: [OpenSIPS-Users] Need some help around script transfermation in
 opensips 3.2 .
In-Reply-To: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
References: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
Message-ID: <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>

Hi,

What is the exact string you are using when getting that err? I'm asking 
as the `Laptop=1;Laptop laptop=1` is invalid due to the space char there 
(a param list does allow such spaces)

Regards

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 24.01.2024 12:04, Sasmita Panda wrote:
> Hi All ,
> I am using opensips 3.2 script transformation parameters to find the 
> existence of a  parameter in  ; separated list of parameters .
>
> In INVITE I have a custom header X-tag which contains a value as below .
> X-Tag : Laptop
>
> Agents used to get registered with X-Tag with some values and I used 
> to put them in the attr column of the location table .
> Agent X-Tag : Laptop=1;Laptop laptop=1
>
> While processing the call , I am matching if the X-Tag in Invite 
> exists in the Agent tag . If the exact parameter exists  then give a 
> call to that contact else not . For that I using below transformation
>
> *{param.exist,Laptop}*
>
> Somehow this is not working . It's giving an error . As for my 
> expectation it should match with the first parameter and as that's an 
> exact match agent must get the call . But its not happening and 
> breaking with error .
>
> *ERROR:core:parse_params: invalid character, ; expected, found l*
> *
> *
> *while checking the existence of parameters, is that only checking a 
> single word without any space ?Also special characters are allowed in 
> between or not  ? *
>
> */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
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From spanda at 3clogic.com  Wed Jan 24 12:53:40 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 24 Jan 2024 18:23:40 +0530
Subject: [OpenSIPS-Users] Need some help around script transfermation in
 opensips 3.2 .
In-Reply-To: <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>
References: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
 <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>
Message-ID: <CALf-Q45bH9gcYm-Z7-ZhV9oN7QBWa8MSkYD+6fSMbvy2vJGzrw@mail.gmail.com>

param list allows special characters like " - or _  "?


*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*


On Wed, Jan 24, 2024 at 6:12 PM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi,
>
> What is the exact string you are using when getting that err? I'm asking
> as the `Laptop=1;Laptop laptop=1` is invalid due to the space char there (a
> param list does allow such spaces)
>
> Regards
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 24.01.2024 12:04, Sasmita Panda wrote:
>
> Hi All ,
> I am using opensips 3.2 script transformation parameters to find the
> existence of a  parameter in  ; separated list of parameters .
>
> In INVITE I have a custom header X-tag which contains a value as below .
> X-Tag : Laptop
>
> Agents used to get registered with X-Tag with some values and I used to
> put them in the attr column of the location table .
> Agent X-Tag : Laptop=1;Laptop laptop=1
>
> While processing the call , I am matching if the X-Tag in Invite exists in
> the Agent tag . If the exact parameter exists  then give a call to that
> contact else not . For that I using below transformation
>
> *{param.exist,Laptop}*
>
>
> Somehow this is not working . It's giving an error . As for my expectation
> it should match with the first parameter and as that's an exact match agent
> must get the call . But its not happening and breaking with error .
>
> *ERROR:core:parse_params: invalid character, ; expected, found l*
>
> *while checking the existence of parameters, is that only checking a
> single word without any space ?Also special characters are allowed in
> between or not  ? *
>
> *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
>
>
>
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From bogdan at opensips.org  Wed Jan 24 12:57:37 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Jan 2024 14:57:37 +0200
Subject: [OpenSIPS-Users] Need some help around script transfermation in
 opensips 3.2 .
In-Reply-To: <CALf-Q45bH9gcYm-Z7-ZhV9oN7QBWa8MSkYD+6fSMbvy2vJGzrw@mail.gmail.com>
References: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
 <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>
 <CALf-Q45bH9gcYm-Z7-ZhV9oN7QBWa8MSkYD+6fSMbvy2vJGzrw@mail.gmail.com>
Message-ID: <57d7a129-9723-4abf-806c-d2487b22627f@opensips.org>

check the grammar as per RFC3261
https://www.ietf.org/rfc/rfc3261.html#section-25.1

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 24.01.2024 14:53, Sasmita Panda wrote:
> param list allows special characters like " - or _ "?
>
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
>
> On Wed, Jan 24, 2024 at 6:12 PM Bogdan-Andrei Iancu 
> <bogdan at opensips.org> wrote:
>
>     Hi,
>
>     What is the exact string you are using when getting that err? I'm
>     asking as the `Laptop=1;Laptop laptop=1` is invalid due to the
>     space char there (a param list does allow such spaces)
>
>     Regards
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com
>        https://www.siphub.com
>
>     On 24.01.2024 12:04, Sasmita Panda wrote:
>>     Hi All ,
>>     I am using opensips 3.2 script transformation parameters to find
>>     the existence of a parameter in  ; separated list of parameters .
>>
>>     In INVITE I have a custom header X-tag which contains a value as
>>     below .
>>     X-Tag : Laptop
>>
>>     Agents used to get registered with X-Tag with some values and I
>>     used to put them in the attr column of the location table .
>>     Agent X-Tag : Laptop=1;Laptop laptop=1
>>
>>     While processing the call , I am matching if the X-Tag in Invite
>>     exists in the Agent tag . If the exact parameter exists  then
>>     give a call to that contact else not . For that I using below
>>     transformation
>>
>>     *{param.exist,Laptop}*
>>
>>     Somehow this is not working . It's giving an error . As for my
>>     expectation it should match with the first parameter and as
>>     that's an exact match agent must get the call . But its not
>>     happening and breaking with error .
>>
>>     *ERROR:core:parse_params: invalid character, ; expected, found l*
>>     *
>>     *
>>     *while checking the existence of parameters, is that only
>>     checking a single word without any space ?Also special characters
>>     are allowed in between or not  ? *
>>
>>     */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
>
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From bogdan at opensips.org  Wed Jan 24 13:10:00 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 24 Jan 2024 15:10:00 +0200
Subject: [OpenSIPS-Users] [RFC] HTTP/2 server support for IMS/5G
Message-ID: <73caf9ee-55c9-4cb0-b23c-2685cba68836@opensips.org>

Hi all,

In the IMS world there is a strong need for OpenSIPS to act as server 
for some non-SIP protocols like DIAMETER - this is already on the 
roadmap, part of tech requirements from the IMS Working Group within 
OpenSIPS project. BTW, Liviu will update on this topic within the next days.

Now, in the 5G standard, HTTP/2 is defined as replacement (want to be) 
for DIAMETER. If acting as HTTP/2 client from within OpenSIPS is rather 
trivial (using the rest_client module), the question that arise here is:
      Does anyone see the need for *HTTP/2 server support*, similar to 
the DIAMETER one ??

Is this need more than a theoretical one, with a practical usage for now 
or near future? if so, is anyone willing to get involved into some 
testing here ?

Best regards,

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
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From a.g at officering.com  Wed Jan 24 14:31:30 2024
From: a.g at officering.com (Alex Goulis)
Date: Wed, 24 Jan 2024 08:31:30 -0600
Subject: [OpenSIPS-Users] [WG-IMS] [RFC] HTTP/2 server support for IMS/5G
In-Reply-To: <73caf9ee-55c9-4cb0-b23c-2685cba68836@opensips.org>
References: <73caf9ee-55c9-4cb0-b23c-2685cba68836@opensips.org>
Message-ID: <CAFyhkvF0=m8J-68St-5-jAJnN64BpE=0TzDwj1q2XwgV5bi90g@mail.gmail.com>

Although I don't see a need to actually replace the diameter you might want
to consider that this is kind of bleeding edge stuff and the more choices
you have for interfacing the better.

The development and maturity of tools available is very fast and fluid and
in the future there might actually be a bona fide need to use the HTTP2

On Wed, Jan 24, 2024, 7:10 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi all,
>
> In the IMS world there is a strong need for OpenSIPS to act as server for
> some non-SIP protocols like DIAMETER - this is already on the roadmap, part
> of tech requirements from the IMS Working Group within OpenSIPS project.
> BTW, Liviu will update on this topic within the next days.
>
> Now, in the 5G standard, HTTP/2 is defined as replacement (want to be) for
> DIAMETER. If acting as HTTP/2 client from within OpenSIPS is rather trivial
> (using the rest_client module), the question that arise here is:
>      Does anyone see the need for *HTTP/2 server support*, similar to the
> DIAMETER one ??
>
> Is this need more than a theoretical one, with a practical usage for now
> or near future? if so, is anyone willing to get involved into some testing
> here ?
>
> 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
>
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From spanda at 3clogic.com  Mon Jan 29 08:47:03 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 29 Jan 2024 14:17:03 +0530
Subject: [OpenSIPS-Users] Need some help around script transfermation in
 opensips 3.2 .
In-Reply-To: <57d7a129-9723-4abf-806c-d2487b22627f@opensips.org>
References: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
 <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>
 <CALf-Q45bH9gcYm-Z7-ZhV9oN7QBWa8MSkYD+6fSMbvy2vJGzrw@mail.gmail.com>
 <57d7a129-9723-4abf-806c-d2487b22627f@opensips.org>
Message-ID: <CALf-Q44MyMEShAjpkBActopkOmR_UFYspT28ybbMC9rHMtgBBQ@mail.gmail.com>

Hi Bogdan ,

I just encountered another discrepancy as below .

In INVITE I have a custom header X-tag which contains a value as below .
X-Tag : *laptop*

Agents used to get registered with X-Tag with some values and I used to put
them in the attr column of the location table .
Agent X-Tag : *Laptop*=1;Product=1


The comparison logic is the same as above  *{param.exist,laptop}  in Agents
attribute . In this case the call is getting processed . I was thinking
this should be a case sensitive match . But it's not . *

*Is this the correct behaviour ? If I need an exact match with case
sensitivity then will that be possible?*

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*


On Wed, Jan 24, 2024 at 6:27 PM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> check the grammar as per RFC3261
>      https://www.ietf.org/rfc/rfc3261.html#section-25.1
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 24.01.2024 14:53, Sasmita Panda wrote:
>
> param list allows special characters like " - or _  "?
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Wed, Jan 24, 2024 at 6:12 PM Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi,
>>
>> What is the exact string you are using when getting that err? I'm asking
>> as the `Laptop=1;Laptop laptop=1` is invalid due to the space char there (a
>> param list does allow such spaces)
>>
>> Regards
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>>   https://www.siphub.com
>>
>> On 24.01.2024 12:04, Sasmita Panda wrote:
>>
>> Hi All ,
>> I am using opensips 3.2 script transformation parameters to find the
>> existence of a  parameter in  ; separated list of parameters .
>>
>> In INVITE I have a custom header X-tag which contains a value as below .
>> X-Tag : Laptop
>>
>> Agents used to get registered with X-Tag with some values and I used to
>> put them in the attr column of the location table .
>> Agent X-Tag : Laptop=1;Laptop laptop=1
>>
>> While processing the call , I am matching if the X-Tag in Invite exists
>> in the Agent tag . If the exact parameter exists  then give a call to that
>> contact else not . For that I using below transformation
>>
>> *{param.exist,Laptop}*
>>
>>
>> Somehow this is not working . It's giving an error . As for my
>> expectation it should match with the first parameter and as that's an exact
>> match agent must get the call . But its not happening and breaking with
>> error .
>>
>> *ERROR:core:parse_params: invalid character, ; expected, found l*
>>
>> *while checking the existence of parameters, is that only checking a
>> single word without any space ?Also special characters are allowed in
>> between or not  ? *
>>
>> *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
>>
>>
>>
>
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From liviu at opensips.org  Mon Jan 29 09:12:39 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 29 Jan 2024 11:12:39 +0200
Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.4 and
	3.2.17 Minor Releases
Message-ID: <43793703-664e-4dff-90fb-86a43998ea52@opensips.org>

Hi, all!

The 3.4.4 and 3.2.17 OpenSIPS minor versions are scheduled for release 
on *Wednesday, Feb 21st*.

In preparation for the releases, starting *Wednesday, Feb 7th*, 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 <https://github.com/OpenSIPS/opensips/issues> that may have 
skipped our attention -- thank you in advance!

PS: we are aware of issue #3286 
<https://github.com/OpenSIPS/opensips/issues/3286>, related to 
rest_client in async mode, and are working towards a resolution in due time.

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com
OpenSIPS Summit 2024 Valencia, May 14-17 |www.opensips.org/events
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From bogdan at opensips.org  Tue Jan 30 07:10:51 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 30 Jan 2024 09:10:51 +0200
Subject: [OpenSIPS-Users] Need some help around script transfermation in
 opensips 3.2 .
In-Reply-To: <CALf-Q44MyMEShAjpkBActopkOmR_UFYspT28ybbMC9rHMtgBBQ@mail.gmail.com>
References: <CALf-Q44fcoHiekexajnaE-ckWoKR4BKgGHYj9NJVrUEZaFQwuQ@mail.gmail.com>
 <287cb3ce-eac2-4d57-a8fb-f7ec181a051e@opensips.org>
 <CALf-Q45bH9gcYm-Z7-ZhV9oN7QBWa8MSkYD+6fSMbvy2vJGzrw@mail.gmail.com>
 <57d7a129-9723-4abf-806c-d2487b22627f@opensips.org>
 <CALf-Q44MyMEShAjpkBActopkOmR_UFYspT28ybbMC9rHMtgBBQ@mail.gmail.com>
Message-ID: <25c2251b-569f-4532-950a-05683519e208@opensips.org>

Hi,

indeed, the checks is case insensitive as the param names in SIP are 
case insensitive (as per RFC).

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com

On 29.01.2024 10:47, Sasmita Panda wrote:
> Hi Bogdan ,
>
> I just encountered another discrepancy as below .
>
> In INVITE I have a custom header X-tag which contains a value as below .
> X-Tag : *laptop*
>
> Agents used to get registered with X-Tag with some values and I used 
> to put them in the attr column of the location table .
> Agent X-Tag : *Laptop*=1;Product=1
>
>
> The comparison logic is the same as above *{param.exist,laptop} in 
> Agents attribute . In this case the call is getting processed . I was 
> thinking this should be a case sensitive match . But it's not . *
> *
> *
> *Is this the correct behaviour ? If I need an exact match with case 
> sensitivity then will that be possible?*
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
>
> On Wed, Jan 24, 2024 at 6:27 PM Bogdan-Andrei Iancu 
> <bogdan at opensips.org> wrote:
>
>     check the grammar as per RFC3261
>     https://www.ietf.org/rfc/rfc3261.html#section-25.1
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com
>        https://www.siphub.com
>
>     On 24.01.2024 14:53, Sasmita Panda wrote:
>>     param list allows special characters like " - or _  "?
>>
>>
>>     */Thanks & Regards/*
>>     /Sasmita Panda/
>>     /Senior Network Testing and Software Engineer/
>>     /3CLogic , ph:07827611765/
>>
>>
>>     On Wed, Jan 24, 2024 at 6:12 PM Bogdan-Andrei Iancu
>>     <bogdan at opensips.org> wrote:
>>
>>         Hi,
>>
>>         What is the exact string you are using when getting that err?
>>         I'm asking as the `Laptop=1;Laptop laptop=1` is invalid due
>>         to the space char there (a param list does allow such spaces)
>>
>>         Regards
>>
>>         Bogdan-Andrei Iancu
>>
>>         OpenSIPS Founder and Developer
>>            https://www.opensips-solutions.com
>>            https://www.siphub.com
>>
>>         On 24.01.2024 12:04, Sasmita Panda wrote:
>>>         Hi All ,
>>>         I am using opensips 3.2 script transformation parameters to
>>>         find the existence of a  parameter in  ; separated list of
>>>         parameters .
>>>
>>>         In INVITE I have a custom header X-tag which contains a
>>>         value as below .
>>>         X-Tag : Laptop
>>>
>>>         Agents used to get registered with X-Tag with some values
>>>         and I used to put them in the attr column of the location
>>>         table .
>>>         Agent X-Tag : Laptop=1;Laptop laptop=1
>>>
>>>         While processing the call , I am matching if the X-Tag in
>>>         Invite exists in the Agent tag . If the exact parameter
>>>         exists  then give a call to that contact else not . For that
>>>         I using below transformation
>>>
>>>         *{param.exist,Laptop}*
>>>
>>>         Somehow this is not working . It's giving an error . As for
>>>         my expectation it should match with the first parameter and
>>>         as that's an exact match agent must get the call . But its
>>>         not happening and breaking with error .
>>>
>>>         *ERROR:core:parse_params: invalid character, ; expected,
>>>         found l*
>>>         *
>>>         *
>>>         *while checking the existence of parameters, is that only
>>>         checking a single word without any space ?Also special
>>>         characters are allowed in between or not  ? *
>>>
>>>         */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
>>
>
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From callum.guy at x-on.co.uk  Tue Jan 30 14:27:41 2024
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Tue, 30 Jan 2024 14:27:41 +0000
Subject: [OpenSIPS-Users] load_balancer usage counter
Message-ID: <CAFjCFznKOYk5QjN1gOze_-mR0H2J7wSHpc0WOi=2Q2hZBUhciA@mail.gmail.com>

Hi All,

I'm implementing the load_balancer module on a very busy system where
thousands of calls may arrive in a matter of seconds. The module is
configured to receive heartbeats every 1 second from many freeswitch
servers, I have this set as a low value to try and keep OpenSIPs up to
date with the real call load, as close to real time as possible.

All servers exist in a single group "channels" to keep the initial
implementation simple. When I kick off the session I use lb_start(1,
"channels", "rs") and if that destination fails I use lb_is_started()
and lb_next() to select the next destination.

With the high call rate I'm concerned that the load values acquired 1
second ago will be used for the entirety of the following second which
would likely lead to a highly imbalanced load. If 1000 calls arrive in
that second I need them spread evenly over the freeswitch servers
which can only happen if OpenSIPs is counting each call as its
allocated.

My hope is that the module increments its counters each time a call is
allocated to a destination; however I have been unable to isolate the
line of code which performs this operation so I'm reaching out for
confirmation. I can see the module adding the dialogs to profiles and
that a separate lb_count_call() method is provided for counting calls
but I'm unclear on the exact usage although it certainly encourages me
that this counting is taking place somewhere.

Thanks for reading,

Callum

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