|Summary:||ASTERISK-14538: One way audio after attended transfer|
|Date Opened:||2009-11-19 11:58:21.000-0600||Date Closed:||2011-09-14 08:51:16|
|Environment:||Attachments:||( 0) 17069.debug.log|
( 1) full2
|Description:||Possibly related to 14249 but I couldn't find a way to reopen that bug, so...|
1. SIP/213 calls SIP/207
2. SIP/207 uses asterisk transfer functionality to transfer to SIP/214
3. SIP/207 hangs up
4. SIP/213 is now bridged to SIP/214
5. Console scrolls with:-
[Nov 19 17:31:59] WARNING chan_sip.c: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x8 (alaw)(8)/0x4 (ulaw)(4)
6. If SIP/214 presses a key the message stops scrolling.
Attached is a noisy (verbose 9 / debug 9 / sip debug) log (on the assumption I can attach a file once I hit submit).
This is with asterisk 1.6.2rc2 (with the patch from issue: 15848) NOT 1.6.2rc6 like it says in the version dropdown - sorry.
|Comments:||By: Paul Belanger (pabelanger) 2010-04-01 13:57:19|
I can confirm this issue with Asterisk 1.4.30.
By: Paul Belanger (pabelanger) 2010-04-01 13:59:53
As a workaround I can set:
and have no issues.
By: Warren Selby (wcselby) 2010-05-28 12:17:46
I was having this issue, I resolved it by removing the "Answer()" from the extension I was transferring to. i.e:
If SIP/2659 answered the original call, then transfered to SIP/2660 without waiting for the person at extension 2660 to answer. Here is the dialplan to explain what I did...
exten => 2660,1,Wait(1)
;exten => 2660,n,Answer ; Commented out this line resolved the issue.
exten => 2660,n,Dial(SIP/2660,6,wW)
exten => 2660,n,Queue(licensing,,,,120)
exten => 2660,n,Voicemail(2660@houston,u)
exten => 2660,n,Hangup
By: Roelof Dijkstra (tallguy74) 2011-04-13 01:02:47
Is anybody still working on this?
I'm also having this issue, and it's still in asterisk version 18.104.22.168.2 (reproduced yesterday 12 april 2011).
To get around this issue i had to go back to asterisk 1.4 (1.4.40), since this seems to be broken in every asterisk 1.6 version so far.
I have multiple SIP connection to different PBX's, and use Asterisk as the proxy for the internal call routing.
If somebody call's from one of the PBX's (A side) to someone on another PBX (B side) true the asterisk proxy, everything works.
If the B side forwards the call to another PBX( C-side) true asterisk and does a blind transfer, it works. However, if you will for the C side to pickup, and then transfer the call, A doesn't hear C while C does hear A.
This prevents us from migrating, and i realy need the TCP SIP...... ;-)
Also, all connections we're tested with one shared codec (ALAW), and also pressing a key doesn't resolve the issue.
By: Russell Bryant (russell) 2011-06-14 19:47:59.345-0500
Is this still a problem? Specifically, is this still an issue with Asterisk 1.8? If you can't reproduce this with 1.8, we will have to close this issue since older versions are no longer supported.
By: Roelof Dijkstra (tallguy74) 2011-08-10 07:33:10.830-0500
I'll try and test the new Asterisk 1.8 version, and see if the problem is still there.
By: Leif Madsen (lmadsen) 2011-09-14 08:51:11.392-0500
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