Your Dial command stops the MOH - if the command were Dial(SIP/1021,20,m) the music would continue until connected or timed-out.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits Sent: Tuesday, August 30, 2011 2:37 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] MOH making calls appear hung up I noticed the CLI shows that the music on hold actually stops for some reason? Here's the output of my CLI: Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363) Verbosity is at least 28 -- Executing [s@ivr-boi-ntc-day:3] Answer("SIP/gw1-000005d6", "") in new stack -- Executing [s@ivr-boi-ntc-day:4] Wait("SIP/gw1-000005d6", "1") in new stack -- Executing [s@ivr-boi-ntc-day:5] Dial("SIP/gw1-000005d6", "SIP/1021,20") in new stack == Using SIP RTP CoS mark 5 -- Called 1021 -- SIP/1021-000005d7 is ringing -- SIP/1021-000005d7 answered SIP/gw1-000005d6 -- Packet2Packet bridging SIP/gw1-000005d6 and SIP/1021-000005d7 -- Started music on hold, class 'default', on SIP/gw1-000005d6 == Using SIP RTP CoS mark 5 -- Executing [6937@from-sip:1] Macro("SIP/1021-000005d8", "stdexten,6937,sip/6937") in new stack -- Executing [s@macro-stdexten:1] Wait("SIP/1021-000005d8", "1") in new stack -- Executing [s@macro-stdexten:2] Dial("SIP/1021-000005d8", "sip/6937,20") in new stack == Using SIP RTP CoS mark 5 -- Called 6937 -- SIP/6937-000005d9 is ringing -- Stopped music on hold on SIP/gw1-000005d6 == Spawn extension (ivr-boi-ntc-day, s, 5) exited non-zero on 'SIP/1021-000005d8<ZOMBIE>' -- Nobody picked up in 20000 ms -- Executing [s@macro-stdexten:3] Goto("SIP/gw1-000005d6", "s-NOANSWER,1") in new stack -- Goto (macro-stdexten,s-NOANSWER,1) -- Executing [s-NOANSWER@macro-stdexten:1] VoiceMail("SIP/gw1-000005d6", "6937,u") in new stack -- <SIP/gw1-000005d6> Playing '/var/spool/asterisk/voicemail/default/6937/unavail.slin' (language 'en') == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on 'SIP/gw1-000005d6' in macro 'stdexten' == Spawn extension (from-sip, 6937, 1) exited non-zero on 'SIP/gw1-000005d6' Thanks! Kevin Oravits From: Danny Nicholas [mailto:da...@debsinc.com] Sent: Tuesday, August 30, 2011 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] MOH making calls appear hung up It seems a reasonable likelihood that your moh at the offending site does not match the codec of the call (IE your MOH is wav and your call codec is SLIN). Set your verbosity and debug up to 15 and try a call to verify this. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Oravits Sent: Tuesday, August 30, 2011 1:53 PM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] MOH making calls appear hung up Greetings, I'm have asterisk servers at about 10 sites, all using Polycom IP 450 phones. With one of my sites, we're having an issue where when a call is transferred, the MOH is not playing and all the caller is hearing is silence. The caller of course thinks they have been hung up on, but the call is actually still in progress and gets successfully transferred if they wait until the person answers. I have researched online and even consulted our 3rd party vendor but no one seems to know how to fix it. Anyone have any advice? Any help would be appreciated. Thanks, Stivaro
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