Summary:	ASTERISK-12760: Audio passed through during dial macro.
Reporter:	theevilapplepie (theevilapplepie)	Labels:
Date Opened:	2008-09-20 17:02:06	Date Closed:	2011-06-07 14:02:55
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Applications/app_dial
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	I set up a dial macro like the privacy manager, it allows the pbx to call a party and give them information about the callee before connecting. The problem is that 7/10 times I call, we can hear one another during the macro. The audio from the macro is still played to the caller only it's slow and garbled.. Dial Statement: Dial(SIP/myoutboundpeer/${mynumber}|30|mM(dialmacro^${TEMPCALLERCID}^argument1^argument2^returncontext^returnextension^returnpriority)) Macro: [macro-dialmacro] exten => s,1,Wait(1) exten => s,2,Read(acceptordecline|./directory/soundfile|1|||10) exten => s,3,GotoIf($[${acceptordecline} = 1]?7) exten => s,4,GotoIf($[${acceptordecline} = 2]?1001) exten => s,5,GotoIf($[${LEN(${acceptordecline})} = 0]?1000) exten => s,6,Goto(2) etc... etc.. The dial macro is huge, I'd rather not post it. Both parties can hear one another and the called side can hear the audio playing. ****** ADDITIONAL INFORMATION ****** Debian 40 R3, Asterisk 1.4.21.2, Zaptel 1.4.12.1 w/ztdummy configured, addons-1.4.7
Comments:	By: theevilapplepie (theevilapplepie) 2008-09-20 17:10:34 Sorry if way I put this is confusing. Someone using the pbx or dialing into the pbx, calls someone who is out of the office. The pbx dials out their cell phone gives them the macro, they then decide based on the caller information if they want to answer or not. It's Debian 4.0 R3. Sorry about the typo. ALSO, this is pure SIP/SIP communication. No analog lines! By: theevilapplepie (theevilapplepie) 2008-09-21 14:45:47 As best I can tell it's the read application that's causing the problem. The audio from both caller and callee are interleaved when the read application starts. I can playback things before the read app but when it hits the problem occurs. By: theevilapplepie (theevilapplepie) 2008-09-21 15:58:41 I successfully had the background application do the same thing. The playback app seems unaffected. By: theevilapplepie (theevilapplepie) 2008-09-21 16:09:14 I upgraded via svn, I downgraded to 1.4.21 and 1.4.20 still same problem. When the macro first starts faint blurbs of audio leak though for the first 2-3 seconds interleaving into the moh for the caller and audio prompts for the callee. If the person in the macro makes consistent loud noise it usually breaks and passes audio both ways and mutes any audio playback in the macro. If both parties are quiet, after 3-5 seconds.. It's completely normal. I can shout and scream and hit digits, no problem, it acts as it should. By: theevilapplepie (theevilapplepie) 2008-09-21 20:03:19 Restarting Zaptel fixes this for a while. I have no physical Zaptel hardware installed and am using Ztdummy. By: Leif Madsen (lmadsen) 2008-12-09 08:15:53.000-0600 Assigned to file to see if he has any ideas right now. Please reassign as necessary should that not be the case. Thanks! By: theevilapplepie (theevilapplepie) 2008-12-11 11:35:51.000-0600 I solved this problem. The problem was caused by my SIP provider sending a re-invite to it's self. I set canreinvite=no on the peer and the problem went away. By: theevilapplepie (theevilapplepie) 2008-12-11 11:43:03.000-0600 It's very analog like though. If both sides are relatively quiet, It barely passes audio. I'm very fond of whistling to make it break through. What will happen is the whistle will slowly burp onto the other side one or twice, then start cutting in and out. Meanwhile the whistler has no audio from the other side.. Then as the interval between the audio cutting in and out gets fewer and fewer eventually it just completely passes audio between BOTH sides. I found this to be very strange and did not think it to be a problem with the sip peer. But after going through all of the sip debug for a call setup, barrier break, and teardown..  I noticed the problem. Unfortunately I was not quick to re-post the solution, so I don't have the SIP debug any longer. Thank you for acknowledging my ticket and I appreciate your help. By: Joshua C. Colp (jcolp) 2008-12-12 09:29:08.000-0600 Closed per reporter.