Interval between Welcome message and onhold prompt

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Interval between Welcome message and onhold prompt

Postby dukemuzaffar » Thu Nov 20, 2014 5:11 am

Hello All,

As soon as the incoming call lands on vicidial, i want to play welcome message, followed by MOH for 15 seconds and then on hold prompt and thereafter MOH and on hold prompt continuously.

But, once after welcome message is played, MOH is playing for 3 sec and soon after that on hold prompt is playing. I want to increase the gap between the welcome message and the on hold prompt and i would like to set that atleast 15 seconds.

Below is the output from CLI.

[Nov 20 14:50:47] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Nov 20 14:50:47] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:47] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:47] -- Playing 'first-1' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:53] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 20 14:50:53] -- Executing [192*168*002*036*1002@default:1] Goto("Local/192*168*002*036*1002@default-0000007e;2", "default,1002,1") in new stack
[Nov 20 14:50:53] -- Goto (default,1002,1)
[Nov 20 14:50:53] -- Executing [1002@default:1] Dial("Local/192*168*002*036*1002@default-0000007e;2", "SIP/1002,60,") in new stack
[Nov 20 14:50:53] == Using SIP RTP CoS mark 5
[Nov 20 14:50:53] -- Called SIP/1002
[Nov 20 14:50:53] -- SIP/1002-0000003c is ringing
[Nov 20 14:50:55] -- Started music on hold, class 'VS_MOH', on SIP/213.132.106.30-0000003b
[Nov 20 14:50:58] -- Stopped music on hold on SIP/213.132.106.30-0000003b
[Nov 20 14:50:58] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:58] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:58] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 20 14:50:58] -- Playing 'second-2' (escape_digits=) (sample_offset 0)

i want MOH to be played for atleast 15 seconds after welcome message and before the on hold prompt.
Is there anyway i can achieve this?

vicidial|asterisk 1.8|vicibox express installation|no cards| getting incoming calls from sip provider through SIP URI
dukemuzaffar
 
Posts: 25
Joined: Wed Oct 22, 2014 6:14 am

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