|Summary:||ASTERISK-04676: Asterisk doesn't signalize end of hold|
|Reporter:||Guenther Starnberger (gst)||Labels:|
|Date Opened:||2005-07-25 16:19:19||Date Closed:||2005-10-02 15:07:14|
|Environment:||Attachments:||( 0) moh.txt|
|Description:||This problem occurs when trying to transfer a call with a Sipura SPA1001 ATA. Both number 1 and number 2 are routed via a PRI/EuroISDN interface (using a TE110P Digium card). I first dial #1. After this #1 answers the call and I put #1 on hold (with the R key). Then I dial #2 and #2 answers. When I hangup the phone on the ATA #1 and #2 are connected together, but #1 thinks that it is still on hold.|
When using 'normal' phones you don't notice that (audio works in both directions). But when #1 is e.g. a mobile it just write 'on hold' on the display (and in my case you hear the MOH of the mobile phone network).
This only happpens when i transfer the #1 and #2 together while #1 is on hold. If I initiate a 3-way-conference by pressing 'R' while I'm connected to #2 and then hangup, #1 and #2 are connected without one of them being on hold.
This bug may or may not be related to the note which I wrote to bug 4756. I had a situation (which I weren't able to reproduce) where I did some transfers between SIP-only phones and at some point phone #3 and #4 were connected together. Phone #3 worked fine, but on #4 I just heard the Asterisk MOH which I have configuried although the audio data from the microphone of #4 was transfered to #3. Please note that in the bug which I am currently reporting Asterisk correctly stopped the MOH, but only the signaling of this on the PRI interface seems to be missing.
I have attached moh.txt which contains the Asterisk output (debug and verbose at 4 - sip debug and pri debug enabled).
|Comments:||By: Olle Johansson (oej) 2005-07-26 00:36:52|
This is related to 3974.
By: Mark Spencer (markster) 2005-08-07 21:02:40
Can you supply a PRI debug as well?
By: Guenther Starnberger (gst) 2005-08-08 01:27:00
The PRI debug is already attached to this report. moh.txt contains the "sip debug" and "pri debug" data.
By: Olle Johansson (oej) 2005-08-11 14:28:53
Check if patch in ASTERISK-3882 changes the situation. Thank you!
By: Guenther Starnberger (gst) 2005-08-11 14:41:04
i've applied reinvite0811.txt from bug ASTERISK-3882 but this doesn't seem to fix the problem. it is still the same behaviour as before.
By: Olle Johansson (oej) 2005-08-11 15:01:44
Thanks for testing.
By: Olle Johansson (oej) 2005-08-25 14:30:42
Ok, I see. I know what this is coming from, it's the who's on hold debate. I will discuss this with Kevin and Mark and sort it out. We can't have one way of handling the AST_CONTROL_HOLD message in chan_sip and another in chan_zap.
By: Michael Jerris (mikej) 2005-09-02 22:35:31
oej- what was the final word on this debate from your time in Alabama?
By: Olle Johansson (oej) 2005-09-06 15:21:56
I don't think we can rewrite the on-hold system for 1.2, since it requires significant changes to Asterisk. Will see if I can find a quick solution.
By: Olle Johansson (oej) 2005-10-02 14:30:09
Please confirm if this is still an issue with latest CVS head code. Thank you for a quick reply.
By: Guenther Starnberger (gst) 2005-10-02 14:56:24
i wasn't able to reproduce the bug. seems to be fixed. tnx.
By: Olle Johansson (oej) 2005-10-02 15:06:59
Fixed in cvs head already.