I'd like to write a brief blog about the status of WebRTC in Debian,
with a focus on SIP
I understand Kamailio 4.0.1 is already in unstable, is that recommended
for potential WebSocket users? Has anybody else written any quickstart
blog about WebRTC with that particular version, possibly with examples
that are consistent with the Debian usage?
For client side, SIPml5 is packaged, and I've had discussions with the
JsSIP guys about packaging.
For TURN, has anybody tried the TURN server project from Google code?
It appears more advanced than the existing two TURN servers in Debian
(e.g. it has database-backed authentication)
http://code.google.com/p/rfc5766-turn-server/
and there is a package in progress:
http://mentors.debian.net/package/rfc5766-turn-server
I'm using an iPhone Bria app and have input Iptel SIP information in the
settings, however, when I try to dial a land line I get a "Forbidden (403)"
error message. What could be wrong?
Hello,
Has something changed about default forking behaviour in >= 4.0?
I have a scenario where INVITEs processed by the proxy first hit a
redirect server, catch a 302, and then append another branch and iterate
over one or more outbound routes.
In the past, this worked fine. After I upgraded to 4.0, I am seeing two
branches at a time on the outbound routes, after the initial branch to
the redirect server. The desired behaviour is serial forking at all times.
tm:failure_reply_mode is set to 3, as it always has been.
Any ideas would be appreciated; thank you!
-- Alex
--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
I can't install siremis 3.0 for kaimailio after install kamailio and begin
to install siremis
in CRIT.log of siremis/log, i
get '05/14/2013','09:13:30','CRIT','ExceptionHandler','cache_dir must be a
directory',''
Anyone can tell me how to do? Thank you all.
2013/5/14 <sr-users-request(a)lists.sip-router.org>
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Future Lian
Hi,
Does kamailio support PRACK method ? Any configuration change is needed?It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow:
A send INVITE --> Kamailio --> proxy the packet to BB send 180 Ringing --> Kamailio -> proxy the packet to AA send PRAK (increase Cseq) --> Kamailio --> proxy the packet to BB send 200 OK --> Kamailio -> proxy the packet to AKamailio re-send 180 Ringing to A
Thanks,AS
Dear List.
I'm trying to issue an automatic unregister() (of a registration) when a
websocket connection is closed. it is useful when the browser is closed
without issuing REGISTER with expires=0.
i've tried implementing this using the event_route[websocket:closed] but i
don't know how to find the registration uri from the $si:$sp. (that is
needed for the unregister() function from the registrar module)
thanks for all replies in advance....
Hello:
I have followed the steps in the guide(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0…:
1. Could you find any mistake in my configuration file?
2. I also wonder how could I be sure of "Be sure you update the listen IP and port as well if Asterisk is running on the same
system with Kamailio" and "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio".
thank you very much
PS:attachment is kamailio、asterisk's congifure file.
Best Regards
zhengyw
----- Original Message -----
From: "Daniel-Constantin Mierla" <miconda(a)gmail.com>
To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>
Sent: Monday, May 13, 2013 3:26 PM
Subject: Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
> Hello,
>
> I am not that familiar to troubleshoot asterisk configuration files, but
> from logs I could see the resulting URI is:
>
> INVITEsip:106@(null) SIP/2.0
>
> That is wrong, meaning something incorrect is done when setting it in
> asterisk. Maybe someone else can help more with asterisk.
>
> Cheers,
> Daniel
>
> On 5/13/13 4:02 AM, zhengyw wrote:
>> hello daniel:
>> thank you very much! but I can't find the problem in the asterisk.
>> attachment is asterisk's configure file, kamailio's configure file and
>> data.can you help with this problem?
>>
>> ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04
>>
>>
>> Best Regards,
>> zhengyw kamailio.cfg
>> <http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg>
>> sip.conf <http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf>
>> extconfig.conf
>> <http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf>
>> extensions.conf
>> <http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf>
>> db_result.txt
>> <http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt>
>>
>>
>>
>> --
>> View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-…
>> Sent from the Users mailing list archive at Nabble.com.
>>
>> _______________________________________________
>> 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
> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
> * http://asipto.com/u/katu *
>
>
> _______________________________________________
> 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
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Hi all,
currently I have been used kamailio and rtpproxy to listen on two different
network (bridge mode), and the register of sip users are saved on different
locations ("aliases" and "location") according the interface where register
is received. When there is a call to a user registered, I use the
registered() function of module registrar on my route[ ] block, to discover
the network where sip user is connected and use the correct parameters in
rtpproxy functions.
>From now, I need that my plataform has forking call support and when the
same sip user is registered on both networks, I have to use the
branch_route[ ] block to process every branch separately because is
necessary different rtpproxy parameters.
I am with difficult in branch_route to set the correct rtpproxy parameters
because I can't find a way to know the network that the branch will be
send. Could anybody tell me if exist a form to solve my problem?
Best Regards
Hello,
I would like to send a SIP NOTIFY message to my SIP client(s), initiated by an event on the server side.
//OFF Asterisk has a similar functionality, for instance ("sip notify <phone vendor>-reboot"), to request a reboot of the phone device from the server side. There is no need for an associated SUBSCRIBE request in such a case. ON//
Our requirement is, that if a specific event happens on the server side, I want to send an event to the client.
My first idea is to use SIP NOTIFY, but perhaps a SIP MESSAGE could do the job as well. The client is registered over TLS, if necessary it could even subscribe for the event.
What would you recommend, using the websockets? Or creating a module from the scratch?
Thank you in advance,
Attila