Hi Mohamed,
Thank you for your interest in helping me,I've configured the the auth_db
module with the Asterisk DB URL and the SIP username and password table
name and verified the MYSQL remote connection from Kamailio to the Asterisk
DB and get connected as predicted.
I tried to register a phone a
looks the issue is only with UDP/TLS/RTP/SAVP re-invite. when i
configured baresip to use RTP/SAVPF, re-invite from baresip to RTP/AVP
sems works fine.
-- juha
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when i make call from UDP/TLS/RTP/SAVP baresip to RTP/AVP sems,
rtpengine gets called on initial invite/200 ok like this and audio works
fine:
Nov 18 02:46:06 rautu /usr/bin/sip-proxy[926]: INFO: =
rtpengine_offer(ICE=force replace-session-connection replace-origin
via-branch=1 RTP/AVP trust
Richard Fuchs writes:
> You only need to specify the transport protocol in an answer if you want
> to change it from whatever the offering client sent (for whatever
> reason). If you don't specify the protocol in the answer (and you
> essentially did that as your flag wasn't understood and ignored
Hi All,
I am testing SIP messages through the proxy from one client to another and
the sender specifies TCP as the transport but Kamailio seems to overwrite
this and chooses UDP (TCP is specified in the contact header).
The receivers are manually registered using kamctl ul add so i am not sure
if
On Tuesday, November 11, 2014 09:29:26 AM Daniel-Constantin Mierla wrote:
> looks like there is a double free, which, although is good to find and
> fix, is normally safe for runtime.
>
> Can you set debug=3 and send all the log messages here?
>
> Cheers,
> Daniel
Daniel, were the logs that we p
Hello,
in the syslog, did you get messages like:
BUG: qm_*: fragm. ... beginning overwritten
Can you send the output of 'bt full'?
Cheers,
Daniel
On 17/11/14 16:49, Kristian Kielhofner wrote:
> Hi Daniel,
>
> It has happened at least twice, which isn't "often" considering the
> scale of our
This is my first time posting to the mailing list. I'm currently running
Kamailio 4.1, per the instructions on the website I got the tar bal for
Siremis 4.1 s well. I have followed the install instructions at
http://kb.asipto.com/siremis:install41x:main and I get all the way down to
the bottom
On 11/17/2014 08:14 AM, Juha Heinanen wrote:
> Juha Heinanen writes:
>
>> rtpengine source code, however, seems to know also these DTLS ones:
>>
>> daemon/call.c: .name = "UDP/TLS/RTP/SAVP",
>> daemon/call.c: .name = "UDP/TLS/RTP/SAVPF",
>>
>> can th
Hey Juha,
Thanks for your promt answer! All clear now.
I just wanted to make sure I optimize my config to maximum (sparing
characters inthere ;)). Maybe someone can update the documentation since
using/not using the params can influence the behaviour.
Cheers,
DanB
if you want to be sure that
Hi Daniel,
It has happened at least twice, which isn't "often" considering the
scale of our deployment but of course it's more often than we'd like
;).
Thanks for looking at this and please let me know what else you need!
On Mon, Nov 17, 2014 at 7:41 AM, Daniel-Constantin Mierla
wrote:
> He
Juha Heinanen writes:
> rtpengine source code, however, seems to know also these DTLS ones:
>
> daemon/call.c:.name = "UDP/TLS/RTP/SAVP",
> daemon/call.c:.name = "UDP/TLS/RTP/SAVPF",
>
> can they be used in rtpengine_offer/answer calls even whe
rtpengine README lists these rtp protocol and profile flags:
+ RTP/AVP, RTP/SAVP, RTP/AVPF, RTP/SAVPF
rtpengine source code, however, seems to know also these DTLS ones:
daemon/call.c: .name = "UDP/TLS/RTP/SAVP",
daemon/call.c: .name = "UDP/TLS/RTP/SAVPF",
Hello,
we did a patch to exec module, but iirc it was after 4.1.6, so it can be
something very old surfaced now.
I will look over it as I get a chance.
Does it happen often or it was just a case?
Cheers,
Daniel
On 15/11/14 23:55, Kristian Kielhofner wrote:
> version: kamailio 4.1.6 (x86_64/lin
Dan Christian Bogos writes:
> I have noticed that if add_contact_alias is called without any
> parameters it will not work (eg: not add the alias parameter in contact)
> in case of ip/port/transport of the received messages are matching what
> the contact declares. On the other hand if I use
>
Hi All,
Looking for some help with a TCP issue in Kamailio.
I am running load testing through the proxy with a topology of UAC > Proxy
> UAS.
With UDP it works fine (using SIPp scripts) - when I turn to TCP as the
transport i get the following error in the kamailio logs:
ERROR: [tcp_main.c:4159
Hey Guys,
For those of you familiar with internal mechanism of add_contact_alias,
I got a question/issue:
I have noticed that if add_contact_alias is called without any
parameters it will not work (eg: not add the alias parameter in contact)
in case of ip/port/transport of the received messa
Hi Dan,
My suggestion will work only if you are routing the call by checking lookup
location. If you are using kamailio as a proxy i am not sure how will you
check the transport protocol.
Can you explain your call routing set in detail ?
Regards
On Fri, Nov 14, 2014 at 4:39 PM, Dan Christian Bo
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