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sdp is in invite and 200 ok. the thing is, nearly the same config still worked in 1.2-branch, but after an upgrade to 1.4 it stopped.
it happens always, no matter if the callee or caller does the reinvite.
the following trace shows what happens. the rtpproxy-port in invite and reply is 41324. they must differ as they are used for different media-streams.
217.10.1.1 - sip-proxy &...
2009-07-01 09:14:12 UTC in Kamailio (OpenSER) SIP server
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happens when i handle a reinvite with rtpproxy_offer("l")
2009-06-30 09:45:13 UTC in Kamailio (OpenSER) SIP server
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ah, sorry, you're right.
2009-06-30 09:41:28 UTC in Kamailio (OpenSER) SIP server
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force_rtp_proxy seems to handle re-invite wrong, resulting in one-way-audio.
2009-06-29 17:29:32 UTC in Kamailio (OpenSER) SIP server
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publishing very long payload-type to the rtp-proxy overflows a buffer. see nathelper.c +2758.
v[1].iov_len must be smaller than sizeof(opts)
2009-06-29 11:55:45 UTC in Kamailio (OpenSER) SIP server
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With this patch conference_play_sound() takes the channels language and passes it to ast_openstream. Also, it adds an AMI-function ConferencePlaySound. The ConferenceSoundComplete event is created with the callers ActionID.
2009-03-17 14:05:54 UTC in AppConference
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Hi,
I noticed that Asterisk renames channels in certain situations. This seems to be ignored by app_conference. The module saves the name of the joining member's channel and does not update it after the renaming. This hurts when a user tries to monitor channels and conference-members because after a renaming, the channelname reported by app_conference is invalid. On the other hand, trying to...
2008-06-23 10:27:38 UTC in AppConference
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Hi Carsten,
I'm afraid but I didn't do any further testing with that patch. IIRC it worked with some other UACs but I am not sure which these were. I had no negative tests. All tested clients worked.
The whole thing didn't go into production, nor was it ported since I was sure it would not make its way into openser and I try to stick to the releases in production.
2008-04-17 07:04:37 UTC in Kamailio (OpenSER) SIP server
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i'd like to suggest something like this:
--- appconference-2.0.1/member.c 2008-02-26 17:05:57.000000000 +0100
+++ appconference-2.0.1-new/member.c 2008-03-25 19:32:45.000000000 +0100
@@ -778,7 +787,11 @@
// int expected_frames = ( int )( floor( (double)( msecdiff( &end, &start ) / AST_CONF_FRAME_INTERVAL ) ) ) ;
// ast_log( AST_CONF_DEBUG, "expected_frames => %d\n"...
2008-04-01 15:45:47 UTC in AppConference
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Hi,
transaction-callbacks are able to change the global transaction-variable (static struct cell *T) e.g. by using t_lookup_ident. This could lead to misbehaviour when tm tries to use T. To solve this issue run_trans_callbacks and run_reqin_callbacks should restore T after running the callback functions.
Regards.
2007-10-04 17:56:43 UTC in Kamailio (OpenSER) SIP server
