Hi,
I'm making my first steps in Kamailio, and need help for a simple
configuration.
I am trying to configure Kamailio to forward SIP traffic from (and to) a
SIP gateway to a Twilio SIP trunk:
SIP gateway ----> Kamailio ----> Twilio SIP trunk
How to configure Kamailio for this use case? How the kamailio.cfg file will
look like in such case?
Thanks.
Hello list,
Hope you all doing well!
We've been attempting to add a URI parameter to implement the trunk group
(tgrp and trunk-context) but discovered the add_uri_param() function only
works with constant string.... we can't use a pseudovar to inform the value
to be added. Anyone knows why such limitation? Can it be changed?
It is possible to do the same thing via the subst_uri function, but it
would be so much more elegant to use the function...
Thank you,
Kind regards,
Patrick Wakano
Hello all.
I have created a WebRTC to SIP gateway. I implemented it using the outbound module. If I restart Kamailio during a call, subsequent messages fail to be routed.
{1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validation
{1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:519]: process_outbound(): failed to decode flow token
{1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:794]: after_loose(): failed to process outbound flow-token
Is it possible to make the flow token survive a restart?
With kind regards
Pan B. Christensen
Developer
Phonect AS
Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway
E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no>
Mobile: 41 88 88 00
[cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/>
[facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>
Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb jenus(a)cyberchaos.nl:
> I do not see any errors in the logs, it looks like it just sets the
> timer to 0 and disconnects the call.
Hello Jan,
I just looked quickly in the code, it seems that the timeout value is not
interpreted as "pseudo-variable", therefore its not possible to use something
like $var in this method right now. It is just read as a numeric value.
dialog.c, line 1342:
if(dlg_set_timeout_by_profile((struct dlg_profile_table *) profile,
&val_s, atoi(timeout_str)) != 0)
This can of course changed in the module with a change in the code.
Best regards,
Henning
--
If you like the work that I do in Kamailio, please consider supporting me on
Patreon: https://www.patreon.com/henningw
From the online documentation:$fU - From URI username
*$fU* - reference to username in URI of 'From' header
It is R/W variable (you can assign values to it directly in configuration
file)
But i am no able to assign values to $fU. Has anyone experienced this?
Snippet of code in the configuration file of mine and its respective logs
showing that $fU is not getting overwritten. I have tried to hard code
values to it assign variables to it, all to no avail.
configuration file code snippet:
xlog("L_INFO", "[$ci] from header username pre mod: $fU");
$fU = "5555555555";
xlog("L_INFO", "[$ci] from header username post mod: $fU");
Logfile snippet:
INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username
pre mod: 3224567880
INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username
post mod: 3224567880
Thanks,
Karthik
Good Morning All!
If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio actually make the SIP call?
At some point I would like outbound calls to be controlled by Kamailio so that the outside endpoints never communicate with the PBX.
Currently a call goes through Kamailio to Asterisk and then Asterisk communicates with the endpoint, and this is night ideal
Thanks ALL
Hi, I have a scenario where I am using.
t_save_lumps(), lookup() and then I can choose to drop some branches in
branch routes.
However, it is possible that I endup without any branch left, in this case
t_relay() is returning and no failure route is called/created.
At this point I need to recover the original message
I believe it is not possible to restore/reset the message without passing
by failure route ?
Else, I could trigger a failure route using t_relay(), is there any
"handy/clean" way to trigger failure_route() ?
Regards
Hello,
I am trying to set up Kamailio as a push notifications proxy, closely
following the example in the "Kamailio in a Mobile World" presentation
(https://www.slideshare.net/FedericoCabiddu/kamailioinamobileworld-51617342).
I am running Debian 9 and Kamailio 5.1.3 from the official Debian
repositories.
I believe the main modules involved in the issue below are tm, tmx, and
tsilo.
Every call passing through the proxy leads to a small memory leak in the tm
module - there is a large amount of "delayed free" memory cells from tm's
internal hash table. At some point the shared memory runs out and Kamailio
restarts. Using the "kamcmd corex.shm_summary" command I was able to see
that the top users of shared memory are "tm: h_table.c: build_cell" and
"core: core/sip_msg_clone.c: sip_msg_shm_clone" with the same allocation
count.
I experimented with removing different parts of the configuration and
noticed that commenting out the "t_continue(...)" call in the "PUSHJOIN"
route
(see slide #22) prevents the leak from happening. Maybe something in that
function is incrementing the reference counter to the hash table cell, but
it is not decrementing the counter when done?
I tried looking around the source code of the tm and tmx modules, but saw
nothing suspicious. I also tried using gdb with a breakpoint in
t_continue_helper (tm/t_suspend.c:166) hoping to see what else is accessing
the htable cell, but was unable to find anything of use.
Has someone encountered anything like this? Can you provide more directions
on debuggin this? I can provide some bits of configuration, but an entire
test setup would be rather difficult, unfortunately.
Thank you for your time,
Ivo
hello dears
i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS , so is there any one has an experiment with this or tested it before ???
if yes, could you please guide me how to do that ? or where to start ?
thanks in advance
Hello dears
when i was trying to build an IMS platform by using kamailio with sip digest authentication scheme and ims_auth module, i couldn't register and the result was as following :
1-the first register message passed to the s-cscf then S-cscf replaying with the message “401 unauthorized - challenging the ue”
2- the second register pass to the S-cscf with the authorization header then the S-CSCF replying with "Forbidden - Private identity not found (Authorization: username)"
So based on that could any one tell me what is the reason of this error and how to fix it ??
Thanks in advance