Hi folks,
I am using Kamailio with IMS, I added the configuration of the WebSocket to the kamailio.cfg in the PCSCF to enable WebRTC clients like (JsSIP) to communicate with each other.. The registration works so far, however when I try to make peer-2-peer call between alice and bob, the caller gets a "403 error" "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" from the S-CSCF..
I did trace on the PCSCF and I found a strange signaling behaviour.
the invite is negotiated between the IMS components as follows:
INVITE from PCSCF to SCSCF
INVITE from PCSCF to PCSCF
PCSCF responses itself with "403 Forbidden - You must register first with a S-CSCF"
SCSCF sends the response "403 Forbidden Mohammed- Dialog not found on S-CSCF or Terminating user not suitable for unregistered services" to the PCSCF
The private IP address of the PCSCF is 172.19.6.35 and of the SCSCF is 172.19.6.36. The PCSCF has a public IP "193.175.132.235"
The IMS components don't run on the same machine, PCSCF and SCSCF are listening to the port 5060. the WebSocket port of the PCSCF is 8000
Can anyone help me get a solution for this problem
Thanks in advance
Best regards,
Medo
Hi,
We are moving away from OpenSIPS and would like to start testing Kamailio.
I really liked how Kamailio complies so clean and is configured easily
in comparison with OpenSIPS monstrosity.
So to not waste time can anybody provide some practical info for...
Kamailio complilation and config examples (w/MySQL) to route Carrier SIP
(DID incoming and outgoing PSTN termination) traffic to and from
Asterisk PBXs.
Example current OpenSIPS setup using dynamic routing module:
1-. dr_rules have the complete DID for PBXs (for incoming traffic from
Carriers to be proxied to the correct PBX).
2-. dr_rules have the partial DID for Carrier gateways (for example
based on internatinal, state, etc. routing of outgoing traffic from PBXs
to Carriers).
3-. dr_gateways have the IP numbers for PBXs.
4-. dr_gateways have the IP numbers for Carriers.
Any info and pointers appreciated.
Best regards,
Gary
Thanks for your response Daniel!
I think I omitted a key piece of information in my original question! We
are using the SCA module.
Hope that sheds some more light on the situation.
-jl
> Date: Thu, 24 Apr 2014 23:06:28 +0200
> From: Daniel-Constantin Mierla <miconda(a)gmail.com>
> To: "Kamailio (SER) - Users Mailing List"
> <sr-users(a)lists.sip-router.org>
> Subject: Re: [SR-Users] Caller ID fields incorrect on Polycom 650s
> Message-ID: <53597CD4.8030106(a)gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
>
> Hello,
>
> kamailio is not touching from/to headers by default. If there are
> incorrect, then could be somewhere else. You ahve tos end a sip trace
> (ngrep for example) with incoming and outgoing sip packets on kamailio
> server.
>
> Cheers,
> Daniel
>
> On 24/04/14 21:58, J Lee wrote:
>> Hi there.
>>
>> I have a situation where the caller id information on my phone appear
>> incorrectly.
>> When I have ext 1 dial ext 2, on the caller's phone, instead of saying
>> "To: 2", it says "From: 2".
>>
>> I'm using kamailio version 4.0.4. I'm just new to everything kamailio
>> and VoIP so I apologize in advance for any remedial questions.
>>
>> So far, I've done a SIP trace to and checked all packets line by line
>> to see if I could spot where the To address was erroneously changed to
>> FROM. In particular, I was checking the call-info fields for caller
>> id data or any other data what would appear on the LED of the phone.
>>
>> Now I'm wondering, who's job (aka which module) is it to set the
>> caller id information for both phones?
>> I'm thinking of digging inside the code for the right module just to
>> see if I can follow what's going on
>>
>> In addition to answering the question above, if you have any
>> suggestions on what else I can check or try, I'd appreciate it.
>>
>> Thanks List!
>>
>>
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users(a)lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
Kamailio SIP Server v4.1.3 stable release is out.
This is a maintenance release of the latest stable branch, 4.1, that
includes fixes since release of v4.1.2. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.1.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.1.3.
For more details about version 4.1.3 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2014/04/kamailio-v4-1-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hi there.
I have a situation where the caller id information on my phone appear
incorrectly.
When I have ext 1 dial ext 2, on the caller's phone, instead of saying "To:
2", it says "From: 2".
I'm using kamailio version 4.0.4. I'm just new to everything kamailio and
VoIP so I apologize in advance for any remedial questions.
So far, I've done a SIP trace to and checked all packets line by line to
see if I could spot where the To address was erroneously changed to FROM.
In particular, I was checking the call-info fields for caller id data or
any other data what would appear on the LED of the phone.
Now I'm wondering, who's job (aka which module) is it to set the caller id
information for both phones?
I'm thinking of digging inside the code for the right module just to see if
I can follow what's going on
In addition to answering the question above, if you have any suggestions on
what else I can check or try, I'd appreciate it.
Thanks List!
Is there a problem with the git repository? When I try to download kamailio (git clone --depth 1
git://git.sip-router.org/sip-router kamailio ) I get an error saying that git.sip-router.org can not be reached. Please let me know, thank you.
Arun
Understand :-) thank you
Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone
-------- Original message --------
From: Carsten Bock
Date:04/24/2014 6:09 AM (GMT-05:00)
To: "Ciprus, Daniel"
Cc: "Kamailio (SER) - Users Mailing List"
Subject: Re: [SR-Users] TCP reset
Hi Daniel,
Define "near future"... :-)
Once we have time, we may look at this topic; but no timeline yet....
Kind regards,
Carsten
2014-04-23 15:55 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>:
> Thanks Carsten,
>
> Any plans to do so in near future ? :-)
>
> On 04/23/2014 09:52 AM, Carsten Bock wrote:
>
> Hi Daniel,
>
> what you want, is basically already implemented in the "Standard"
> usrloc module, see:
>
> http://kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.handle_…
>
> We haven't implemented that feature yet in the IMS modules....
>
> Schöne Grüße,
> Carsten
>
>
> 2014-04-23 15:47 GMT+02:00 Daniel Ciprus <daniel.ciprus(a)acision.com>:
>
> Hi,
>
> Is there any way to remove registration entry from registrar (Kamailio IMS
> built on stable 4.1) detected on p-cscf by incoming TCP RST from the network
> ? What's happening is that client is loosing connections/crashing and TCP
> RST is sent back to pcscf which is not propagating changes back to
> registrar. This is likely not possible with my version of kamailio but at
> least logging which would indicate change of the status will be helpful.
>
> thanks for any hints
>
>
> --
> Daniel Ciprus
> Integration engineer
> http://www.acision.com
>
> 9954 Mayland Dr
> Suite 3100
> Richmond, VA 23233
> USA
> T: +1 804 762 5601
> E: daniel.ciprus(a)acision.com
>
> ________________________________
> This e-mail and any attachment is for authorised use by the intended
> recipient(s) only. It may contain proprietary material, confidential
> information and/or be subject to legal privilege. It should not be copied,
> disclosed to, retained or used by, any other party. If you are not an
> intended recipient then please promptly delete this e-mail and any
> attachment and all copies and inform the sender. Thank you for
> understanding.
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> --
> Daniel Ciprus
> Integration engineer
> http://www.acision.com
>
> 9954 Mayland Dr
> Suite 3100
> Richmond, VA 23233
> USA
> T: +1 804 762 5601
> E: daniel.ciprus(a)acision.com
>
> ________________________________
> This e-mail and any attachment is for authorised use by the intended
> recipient(s) only. It may contain proprietary material, confidential
> information and/or be subject to legal privilege. It should not be copied,
> disclosed to, retained or used by, any other party. If you are not an
> intended recipient then please promptly delete this e-mail and any
> attachment and all copies and inform the sender. Thank you for
> understanding.
>
--
Carsten Bock
CEO (Geschäftsführer)
ng-voice GmbH
Schomburgstr. 80
D-22767 Hamburg / Germany
http://www.ng-voice.com
mailto:carsten@ng-voice.com
Office +49 40 34927219
Fax +49 40 34927220
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
________________________________
This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.
Hi,
Is there any way to remove registration entry from registrar (Kamailio IMS built on stable 4.1) detected on p-cscf by incoming TCP RST from the network ? What's happening is that client is loosing connections/crashing and TCP RST is sent back to pcscf which is not propagating changes back to registrar. This is likely not possible with my version of kamailio but at least logging which would indicate change of the status will be helpful.
thanks for any hints
--
Daniel Ciprus
Integration engineer
http://www.acision.com
9954 Mayland Dr
Suite 3100
Richmond, VA 23233
USA
T: +1 804 762 5601
E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com>
________________________________
This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.
Hello,
I am considering releasing v4.1.3 by mid of next week, on Wednesday or
Thursday (April 23 or 24). If there are issues you are aware of and not
reported to the bug tracker, add them there asap to investigate them.
Also, if you noticed some fixes in the master branch not backported
yet, report them to the mailing lists.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
