Summary:ASTERISK-23541: Asterisk 12.1.0 Not respecting directmedia=no and issuing REINVITE
Reporter:Justin E (devlap)Labels:
Date Opened:2014-03-26 14:05:14Date Closed:2014-06-01 10:32:51
Status:Closed/CompleteComponents:Bridges/bridge_native_rtp Channels/chan_sip/General
Versions:12.1.0 Frequency of
Environment:Centos 6.5Attachments:( 0) call.log
( 1) sipconf.txt
( 2) testcall.txt
Description:Looks like * 12.1.0 is not respecting disabling re-invites (directmedia=no)
--- Debug : https://db.tt/S9DyclDr
--- Config:  http://pastebin.com/dqA3MTY8
Comments:By: Matt Jordan (mjordan) 2014-03-26 15:44:27.116-0500

pastebins have a tendency to die. Please attach the logs and configs to this issue as text files.

By: Justin E (devlap) 2014-03-26 18:22:32.069-0500

Uploaded the sip.conf and Sip Debug/Verbose 10 console log of a call.

By: Justin E (devlap) 2014-03-27 09:40:48.251-0500

Just verified the logs and config, it is all as posted.

By: Rusty Newton (rnewton) 2014-04-07 13:32:29.351-0500

Your log doesn't include DEBUG or VERBOSE type messages. You may have turned up the log levels, but you'll need to verify they show up in the log. You can use logger.conf to turn things on, and then "logger show channels" and "core show settings" to see what types of messages are being written where and what levels they are at during runtime.


Additionally, if you can attach a pcap that matches the trace in the Asterisk log file that is extremely helpful as we can view that with wireshark which makes everything easier. However the pcap doesn't let us see the important log messages in the Asterisk logs.

By: Justin E (devlap) 2014-04-08 18:43:58.391-0500

Attached is a debug log of the call. I followed the documentation on generating it.

By: Rusty Newton (rnewton) 2014-04-14 09:51:05.785-0500

It is not clear what is going on and I am unable to reproduce this.

Can you provide the dialplan used for the call origination, plus another debug log, but with a packet capture to match it. You'll also need to point out which call the issue occurs on; that is, provide the call-ID to be clear.

Does the issue occur 100% of the time when using the sip.conf configuration you posted?

By: Matt Jordan (mjordan) 2014-04-29 15:55:24.322-0500

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.  Further information can be found at http://www.asterisk.org/developers/bug-guidelines

By: Matt Jordan (mjordan) 2014-04-29 16:23:27.140-0500

Re-opening per conversation in #asterisk-bugs

By: Justin E (devlap) 2014-04-29 16:31:07.113-0500

RE: The IRC Chat:

" Im guessing the issue is : [Apr  8 16:37:18] DEBUG[13070][C-0000015d] chan_sip.c: Sending reinvite on SIP '121ad393320420de7276f4cc42b90482@' - It's audio soon redirected to IP "

" is the asterisk box on the lan side. the originating phone was"
My SIP Provider is on and my wan to them I believe is

By: Justin E (devlap) 2014-04-29 16:32:30.315-0500

Basically what is occuring is if I  make a call one side can't hear the other.

I believe this is because the outside line is being invited to speak directly to my phones on and this is an internal lan and not addressable by the other side.

I was told that directmedia=no resolves this and routes everything VIA the asterisk box, but it appears that a REINVITE is being issued anyways.