Call hangup after resume

[hassan.ali]

Hello,

My vici details are:
Vicibox Redux v.5.0.2-130821
Asterisk version: 1.8.31.0

Server Details:
Single server | 16GB RAM | 350 GB HDD | No additional hardware

My outbound calls are working fine.

But when i dial an inbound call, it displays call in queue on agent screen, i can listen music on hold as i change my status from pause to resume call handup

my CLI shows this:

[May 25 02:34:59] == Using SIP RTP CoS mark 5
[May 25 02:34:59] -- Executing [0092532210050@mppl_extension:1] Answer("SIP/mppl-trunk-0000002e", "") in new stack
[May 25 02:34:59] -- Executing [0092532210050@mppl_extension:2] AGI("SIP/mppl-trunk-0000002e", "agi-DID_route.agi") in new stack
[May 25 02:34:59] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[May 25 02:34:59] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20160525023459_0092532210050_00923018114401)
[May 25 02:34:59] -- <SIP/mppl-trunk-0000002e>AGI Script agi-DID_route.agi completed, returning 0
[May 25 02:34:59] -- Executing [99909*1***DID@default:1] Answer("SIP/mppl-trunk-0000002e", "") in new stack
[May 25 02:34:59] -- Executing [99909*1***DID@default:2] AGI("SIP/mppl-trunk-0000002e", "agi-VDAD_ALL_inbound.agi") in new stack
[May 25 02:34:59] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[May 25 02:35:00] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:00] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:00] WARNING[13021]: file.c:666 ast_openstream_full: File http//172.16.0.223/RECORDINGS/MP3/20150303-104403_03028114402-all.mp3 does not exist in any

format
[May 25 02:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 25 02:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 25 02:35:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 25 02:35:01] -- Started music on hold, class 'default', on SIP/mppl-trunk-0000002e
[May 25 02:35:03] == Manager 'sendcron' logged off from 127.0.0.1
[May 25 02:35:04] -- Stopped music on hold on SIP/mppl-trunk-0000002e
[May 25 02:35:04] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:04] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:04] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:04] WARNING[13021]: file.c:666 ast_openstream_full: File http//172.16.0.223/RECORDINGS/MP3/20150303-104403_03028114402-all.mp3 does not exist in any

format
[May 25 02:35:05] -- Started music on hold, class 'default', on SIP/mppl-trunk-0000002e
[May 25 02:35:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 25 02:35:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 25 02:35:12] == Manager 'sendcron' logged on from 127.0.0.1
[May 25 02:35:12] -- Executing [172*016*000*223*78600051@default:1] AGI("Local/172*016*000*223*78600051@default-00000011;2", "agi://127.0.0.1:4577/call_log") in

new stack
[May 25 02:35:12] -- <Local/172*016*000*223*78600051@default-00000011;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 25 02:35:12] -- Executing [172*016*000*223*78600051@default:2] Dial("Local/172*016*000*223*78600051@default-00000011;2",

"SIP/trunk1/172*016*000*223*78600051,,tTo") in new stack
[May 25 02:35:12] == Using SIP RTP CoS mark 5
[May 25 02:35:12] -- Called SIP/trunk1/172*016*000*223*78600051
[May 25 02:35:12] == Spawn extension (default, 172*016*000*223*78600051, 2) exited non-zero on 'Local/172*016*000*223*78600051@default-00000011;2'
[May 25 02:35:12] -- Executing [h@default:1] AGI("Local/172*016*000*223*78600051@default-00000011;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----

NODEBUG-----0-----CANCEL----------") in new stack
[May 25 02:35:12] -- <Local/172*016*000*223*78600051@default-00000011;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----0-----

CANCEL---------- completed, returning 0
[May 25 02:35:12] -- Stopped music on hold on SIP/mppl-trunk-0000002e
[May 25 02:35:12] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:12] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:13] == Manager 'sendcron' logged off from 127.0.0.1
[May 25 02:35:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 25 02:35:13] -- <SIP/mppl-trunk-0000002e>AGI Script agi-VDAD_ALL_inbound.agi completed, returning -1
[May 25 02:35:13] -- Executing [172*016*000*223*8600051@default:1] AGI("SIP/mppl-trunk-0000002e", "agi://127.0.0.1:4577/call_log") in new stack
[May 25 02:35:13] -- <SIP/mppl-trunk-0000002e>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 25 02:35:13] -- Executing [172*016*000*223*8600051@default:2] Dial("SIP/mppl-trunk-0000002e", "SIP/trunk1/172*016*000*223*8600051,,tTo") in new stack
[May 25 02:35:13] == Using SIP RTP CoS mark 5
[May 25 02:35:13] -- Called SIP/trunk1/172*016*000*223*8600051
[May 25 02:35:14] == Everyone is busy/congested at this time (1:0/0/1)
[May 25 02:35:14] -- Executing [172*016*000*223*8600051@default:3] Hangup("SIP/mppl-trunk-0000002e", "") in new stack
[May 25 02:35:14] == Spawn extension (default, 172*016*000*223*8600051, 3) exited non-zero on 'SIP/mppl-trunk-0000002e'
[May 25 02:35:14] -- Executing [h@default:1] AGI("SIP/mppl-trunk-0000002e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----

CHANUNAVAIL----------") in new stack
[May 25 02:35:14] -- <SIP/mppl-trunk-0000002e>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed,

returning 0
[May 25 02:35:16] == Manager 'sendcron' logged on from 127.0.0.1
[May 25 02:35:16] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-00000012;2", "8600051,Fmq") in new stack
[May 25 02:35:16] > Channel Local/58600051@default-00000012;1 was answered.
[May 25 02:35:16] -- Executing [8309@default:1] AGI("Local/58600051@default-00000012;1", "agi://127.0.0.1:4577/call_log") in new stack
[May 25 02:35:16] -- <Local/58600051@default-00000012;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 25 02:35:16] -- Executing [8309@default:2] Dial("Local/58600051@default-00000012;1", "SIP/trunk1/8309,,tTo") in new stack
[May 25 02:35:16] == Using SIP RTP CoS mark 5
[May 25 02:35:16] -- Called SIP/trunk1/8309
[May 25 02:35:16] == Everyone is busy/congested at this time (1:0/0/1)
[May 25 02:35:16] -- Executing [8309@default:3] Hangup("Local/58600051@default-00000012;1", "") in new stack
[May 25 02:35:16] == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-00000012;1'
[May 25 02:35:16] -- Executing [h@default:1] AGI("Local/58600051@default-00000012;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----

CHANUNAVAIL----------") in new stack
[May 25 02:35:16] -- <Local/58600051@default-00000012;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------

completed, returning 0
[May 25 02:35:16] == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-00000012;2'
[May 25 02:35:16] -- Executing [h@default:1] AGI("Local/58600051@default-00000012;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0

---------------") in new stack
[May 25 02:35:16] -- <Local/58600051@default-00000012;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed,

returning 0

Please help me to sort this out.

[hassan.ali]

Anyone please help me out. I am waiting for reply.

[williamconley]

1) You listed your installer version (very good), but left off your Vicidial Version with Build. This is a requirement for Free Support on this board.

2) Never reply to your own post. All you've done is remove your post from the list of "not answered" posts on the system, thus causing many of us to ignore you because we believe someone else responded to your post. But it was you trying to bump yourself to the top of the list, while actually pushing yourself OFF the list! LOL

3) Try SIP debug and consider disallow=all/allow=ulaw for your sip account entry for the inbound call. It is entirely possible that you've selected "all" and the carrier is attempting g729, but you don't have g729 so the call fails. When you say "all" instead of listing your actual available codecs, you set yourself up for a fail when the call attempts to transmit sound. (This was just a guess, of course, based on previous experience.)

4) Was the agent in question "OnHook" or regular logged in?

[hassan.ali]

Thanks for your reply, it is my first post and i'll keep your suggestions in mind while attempting my second.

1) VERSION: 2.10-442a BUILD: 140612-2153

3) I have tried codecs all the ways but it was not the issue i think

4) Regular logged in agents

[williamconley]

You're using an old version that didn't get a lot of debugging. You may want to upgrade to the latest version to avoid any possible oddity (the present version has had a lot more debugging, and has a lot of present users!).

To upgrade follow the upgrade instructions in the wiki (or cd /usr/src/astguiclient/trunk; svn up; and then follow the instructions in the UPGRADE document in that folder).

You should also try sip debug as I suggested earlier.

Code: Select all

sip set debug ip xxx.xxx.xxx.xxx

This command will limit the debug text to that single IP instead of all live calls.