[asterisk-users] MOH making calls appear hung up

[Danny Nicholas]

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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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 at ivr-boi-ntc-day:3] Answer("SIP/gw1-000005d6", "") in new
stack

    -- Executing [s at ivr-boi-ntc-day:4] Wait("SIP/gw1-000005d6", "1") in new
stack

    -- Executing [s at 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 at from-sip:1] Macro("SIP/1021-000005d8",
"stdexten,6937,sip/6937") in new stack

    -- Executing [s at macro-stdexten:1] Wait("SIP/1021-000005d8", "1") in new
stack

    -- Executing [s at 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 at macro-stdexten:3] Goto("SIP/gw1-000005d6",
"s-NOANSWER,1") in new stack

    -- Goto (macro-stdexten,s-NOANSWER,1)

    -- Executing [s-NOANSWER at 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:danny at 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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Oravits
Sent: Tuesday, August 30, 2011 1:53 PM
To: 'asterisk-users at 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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