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Summary:ASTERISK-10184: Attended transfer leaves two sides without audio path
Reporter:Yehavi Bourvine (yehavi)Labels:
Date Opened:2007-08-28 06:43:00Date Closed:2011-06-07 14:07:48
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Transfers
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) test_81632.txt
( 1) test.cap
( 2) test.txt
Description:When doing an attended transfer, after the middle party sends the "refer" packet, Asterisk does not send any INVITE to the other parties so they can talk now one to the other.

****** ADDITIONAL INFORMATION ******

In the trace bellow:
7106 calls 89444; they talk and hear each other.
89444 puts 7106 on hold and calls 80620; it answers and they can talk.
89444 press the "transfer" causing it to send a REFER message.

Asterisk does not send any re-INVITE to 7106 & 80620 thus they have no audio path.

The problem happend with "vanilla" asterisk as well as with patch 9305 applied.
Comments:By: Russell Bryant (russell) 2007-08-28 10:09:41

Try the latest code.  A patch just went in for this in rev 81210.
By: Yehavi Bourvine (yehavi) 2007-08-29 04:43:36

I'll check it tomorrow and send a feedback.

               Thanks, __Yehavi:
By: Yehavi Bourvine (yehavi) 2007-08-30 02:26:59

Sorry, still no luck...
Tried with revision 81365 (fresh download from today) and the same behaviour.

                               Thanks, __Yehavi:
By: Yehavi Bourvine (yehavi) 2007-09-06 00:07:41

I've tried again with SVN 81632 (files test_81632.txt and test.cap). It now sends the INVITEs, but if I understand correctly it sends the incorrect SDP to each side (i.e. instead of sending SDP of 1 to station 2 it sends SDP of 1 to station 1 and vice versa).

                    Thanks! __Yehavi:
By: Yehavi Bourvine (yehavi) 2007-09-18 06:24:19

Tried again today with r82728; we are back to square one: no Invites are sent after the REFER.

                            Thanks, __Yehavi:
By: Yehavi Bourvine (yehavi) 2007-10-17 01:59:48

The problem still exists on 1.4.13; am I the only one having this problem???

                             Thanks! __Yehavi:
By: Yehavi Bourvine (yehavi) 2007-10-21 06:16:58

I've moved the 1.4.13 version from my lab to the production system (where I have more phone types than  the lab) and it seems that the problem happens only with SNOM phones, while Cisco & Polycom works ok.

Since I have only SNOM in my lab I did not notice it. Seems like this bug can be closed.

                      Thanks! __Yehavi: