On Sunday 30 Oct 2011, Sammy Govind wrote: > Try turning on the Sip debug for the PSTN call as well as RTP debug. > Paste the output here.
There's no SIP involved until the call is picked up -- only PSTN and IAX2. No RTP log available either. > > The Dial server is connected to multiple 4-port Redfone devices for > > handling PSTN incoming and outgoing calls. Outgoing calls always > > originate from and incoming calls always terminate at the SIP > > server. SIP and Dial servers are connected over IAX2. > > Explain the above abit as well..couldnt get the clear picture of what > it looks like. Seems to me that you guys are using two servers and > call-audio gets lost in between the servers OR in between the > Dial-Server and redfone device for Queue Calls. After looking further, the problem seems to be purely in playing recorded messages over IAX2. Looking at the debug logs on the SIP server (which is playing the recorded messages) shows that it stops playing one of the messages at some point in the flow, and then never plays anything again. Here's the logs from three different calls: *** Announcement gets cut off in the "please hold" message > requested format = alaw, > requested prefs = (alaw|gsm|ulaw), > actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [6000@cg-20:19] NoOp("IAX2/dialbank-1-5531", "Spygroup done") in new stack -- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-5531", "") in new stack -- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-5531", "1") in new stack -- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-5531", "followme/pls-hold- while-try") in new stack -- <IAX2/dialbank-1-5531> Playing 'followme/pls-hold-while-try.slin' (language 'en') *** Caller hung up after some time of silence == Spawn extension (cg-20, 6000, 22) exited non-zero on 'IAX2/dialbank-1-5531' -- Hungup 'IAX2/dialbank-1-5531' *** Announcement gets cut off in the first *** "Your call is now next in line" message -- Accepting AUTHENTICATED call from 10.0.10.132: > requested format = alaw, > requested prefs = (alaw|gsm|ulaw), > actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-4966", "") in new stack -- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-4966", "1") in new stack -- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-4966", "followme/pls-hold- while-try") in new stack -- <IAX2/dialbank-1-4966> Playing 'followme/pls-hold-while-try.slin' (language 'en') -- Executing [6000@cg-20:23] Wait("IAX2/dialbank-1-4966", "1") in new stack -- Executing [6000@cg-20:32] Queue("IAX2/dialbank-1-4966", "cg-20,dr,,,,,,,,") in new stack -- <IAX2/dialbank-1-4966> Playing 'queue-youarenext.slin' (language 'en') *** Caller hung up after some time of silence == Spawn extension (cg-20, 6000, 32) exited non-zero on 'IAX2/dialbank-1-4966' -- Hungup 'IAX2/dialbank-1-4966' *** Caller hears "please hold", "Your call is now next in line", *** "thank you for your patience" messages, a few rings, *** then sound gets cut off in the next "Your call is now next in line" *** message -- Accepting AUTHENTICATED call from 10.0.10.132: > requested format = alaw, > requested prefs = (alaw|gsm|ulaw), > actual format = alaw, > host prefs = (alaw), > priority = mine -- Executing [6000@cg-20:20] Answer("IAX2/dialbank-1-2655", "") in new stack -- Executing [6000@cg-20:21] Wait("IAX2/dialbank-1-2655", "1") in new stack -- Executing [6000@cg-20:22] BackGround("IAX2/dialbank-1-2655", "followme/pls-hold- while-try") in new stack -- <IAX2/dialbank-1-2655> Playing 'followme/pls-hold-while-try.slin' (language 'en') -- Executing [6000@cg-20:23] Wait("IAX2/dialbank-1-2655", "1") in new stack -- Executing [6000@cg-20:32] Queue("IAX2/dialbank-1-2655", "cg-20,dr,,,,,,,,") in new stack -- <IAX2/dialbank-1-2655> Playing 'queue-youarenext.slin' (language 'en') -- Told IAX2/dialbank-1-2655 in cg-20 their queue position (which was 1) -- <IAX2/dialbank-1-2655> Playing 'queue-thankyou.slin' (language 'en') -- <IAX2/dialbank-1-2655> Playing 'queue-youarenext.slin' (language 'en') *** Caller hung up after some time of silence == Spawn extension (cg-20, 6000, 32) exited non-zero on 'IAX2/dialbank-1-2655' -- Hungup 'IAX2/dialbank-1-2655' All this works perfectly when tested from a client logged in to the SIP server itself. The problem only arises when a caller calls the Dial server from the PSTN, which then passes the call to the SIP server over IAX2. Regards, -- Raj -- Raj Mathur r...@kandalaya.org http://kandalaya.org/ GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F PsyTrance & Chill: http://schizoid.in/ || It is the mind that moves -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users