Ok so
It now includes the record-route but its still not modifying the
contact header and the problem persists.
¿Any suggestions as to how to do that?
Cheers
David
On 2 September 2011 21:17, David Zambrano wrote:
> Thanks Andrew
> Ill deal with the ping config once I resolve the route probl
Hello,
Thank you very much, I will study this I proposed :-).
Regards
2011/9/5 Daniel-Constantin Mierla :
> Hello,
>
> On 9/2/11 8:40 PM, Anto wrote:
>>
>> Hello
>>
>> I want to store in a file access and disconnections to the system. To
>> access (registrar), I thought scheduling a perl script
Hi!
Recently I had a problem with permanent usrloc entries when upgrading a
server from SER 0.9 to Kamailio 3.1.
Obviously the problem was that the permanent entries had "flag=128" and
now expires=0 is used, see old thread:
http://old.nabble.com/usrloc-permanent-records-and-strptime-td17668417.h
It seems that something is miss configured on your server. The fixes
that I made in the trunk (and you pulled in your local 3.1 repo) were
designed to handle the scenario that you are trying to implement.
The ACK should be handled properly and routed to the upstream carrier
(following the same pat
Hi,
I am using the XCAP server of kamailio for contact list and presence
rules. It seems it only support to update the whole xcap doc of the
user but not the partial update. Can anyone tell me if the kamailio's
xcap server support the partial update of a user's doc? for example,
to remove a contact
Hello,
can you use t_flush_flags() after setting the accounting flag in
falure_route? Automatic update was missing so far, reported by Alex
Hermann as well. I just did a patch, so if you want to try it, see the
commit:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c589c
I have no idea how siptrace works, but: When using tm module, the CANCEL
is not forwarded but absorbed by tm module, and then tm module creates a
new CANCEL based on the previous sent INVITE.
That may be the reason why on_send route is not activated.
klaus
Am 06.09.2011 10:24, schrieb Timo Klec
Make sure force_rtpproxy is called also in reply route, and make sure
the reply route is activated (t_on_reply).
See the default config file for how it works.
For debugging use (or Wireshark ...)
ngrep -i any -W byline port 5060
and analyze the SDP before and after Kamailio.
Also, add xlogl()
Hello all,
I want to log the incoming and outgoing CANCEL on my Kamailio. I tried with
a call of sip_trace in the default route
setflag(22);
if (method == "CANCEL") {
sip_trace();
if (t_check_trans()) {
t_relay();
Hello,
On 9/5/11 4:55 PM, MÉSZÁROS Mihály wrote:
Hello Daniel,
Thank You, and many thanks to Alex too!
The fix you made is working fine for me.
thanks for testing.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipt
Hi
I want to use Kamailio to handle both signalling and media always.
My scenario is the following PSTN==>KAMAILIO==>ASTERISK. I have a dialplan
that points to ASTERISK when a call comes into Kamailio. Signalling is fine,
however i cant seem to get the Media to work, i have installed the RTPPROXY
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