Dear All,
I'm using kamailio as Proxy server for registration with RTP proxy as relay
server for voice packets...I have an issue that when sending an Invite, the
audio address in SDP (o header)is not RTP Proxy address while if I send
another Invite to kamailio (second Invite), the RTP Proxy
Yes, correct. I don't think this is anything to do with Kamailio, I will detach
you from the thread,
Thanks for all your assistance!
Robert Jongbloed
Vox Lucida Pty. Ltd.
On 12/06/2011, at 8:11 PM, Iñaki Baz Castillo wrote:
The id of both tuples are different so the server must
What is the IP address if the first case? A wrong one or the original IP
address provided by the caller?
If it is the original IP address, either force_rtp_proxy was not called
or there was a problem wiht the rtpproxy.
Use xlog statements in front and after the rtpproxy functions and verify
Hello,
On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
Hello.
DCM Hello,
DCM to understand the scenario:
DCM - first branch has destination uri ($du) set
Exactly.
DCM - it failed and gets to failure route where you call ds_next_domain()
DCM and $du s still the one from first branch?
Dear Sir,
Can you please have a look to my config file and let me know please why the
rtp proxy is not forced in SDP?
PasteBin address:
http://pastebin.com/NrJR8geQ
Regards
On Mon, Jun 13, 2011 at 10:49 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
What is the IP address if the
On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
Hello,
On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
Hello.
DCM Hello,
DCM to understand the scenario:
DCM - first branch has destination uri ($du) set
Exactly.
DCM - it failed and gets to failure route
Hello,
On 6/11/11 9:24 AM, Iñaki Baz Castillo wrote:
2011/6/10 Iñaki Baz Castilloi...@aliax.net:
Then if you use a simple config with t_relay(), go to tm module, see where
t_relay() is defined, follow a bit the code and set a breakpoint by file and
line number at a convenient location (as much
On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
Hello,
On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
Hello.
DCM Hello,
DCM to understand the scenario:
DCM - first branch has destination uri ($du) set
Exactly.
DCM - it
Hello,
On 6/11/11 6:54 AM, Iñaki Baz Castillo wrote:
2011/6/10 Daniel-Constantin Mierlamico...@gmail.com:
have you spotted in SIP/SIMPLE specs what has to be sent when the presentity
is offline? My quick google was not that succesful. Maybe Inaki has the
specs more fresh indexed in memory and
On Mon, 2011-06-13 at 12:12 +0200, Daniel-Constantin Mierla wrote:
On 6/13/11 12:03 PM, Evgeniy Spinov wrote:
On Mon, 2011-06-13 at 10:49 +0200, Daniel-Constantin Mierla wrote:
Hello,
On 6/12/11 9:56 PM, Spinov Evgeniy wrote:
Hello.
DCM Hello,
DCM to understand the
2011/6/13 Daniel-Constantin Mierla mico...@gmail.com:
I saw Andrei jumped in and added case insensitive comparison of naptr flags
-- just to conclude this discussion, is it working on now?
Hi Daniel, I've tested it right *now* :)
Yes, it works.
--
Iñaki Baz Castillo
i...@aliax.net
Dear All,
I'm using the below config plan for routing my calls...The issue that
forcing rtp proxy is not working well and the rtp proxy is never
forcedCan you please provide me a guidance here?
if($rU=~^00.* )
{
if(!cr_route(default, domain.com, $rU, $rU,
On Jun 13, 2011, at 2:48 PM, Iñaki Baz Castillo i...@aliax.net wrote:
2011/6/13 Daniel-Constantin Mierla mico...@gmail.com:
I saw Andrei jumped in and added case insensitive comparison of naptr flags
-- just to conclude this discussion, is it working on now?
Hi Daniel, I've tested it
It does not work, t_newtran always returns success, so it will never
absorb the retransmission.
So what I did was:
if ($sht(a=$ci::retrans) == 0) {
$sht(a=$ci::retrans) = 1;
} else {
exit();
}
xlog(L_INFO,XLOG: xxx);
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