Inbound call is getting disconnected as soon as I answer

[tanvirvalolok]

I have 3 Remote (On-Hook) agents setup for DID through a Campaign.

It's supposed to ring all my phones at a same time when somebody calls the DID.
Its working fine till it rings all phones together, but as soon as I answer the call on any of those phones, instead of connecting the call gets disconnected on that phone.

Code: Select all

Connected to Asterisk 13.29.2-vici currently running on 68-168-96-145 (pid = 22650)
[Dec 22 09:17:56]   == Using SIP RTP CoS mark 5
[Dec 22 09:17:56]        > 0x7f1234051340 -- Strict RTP learning after remote address set to: 64.125.111.109:26592
[Dec 22 09:17:56]     -- Executing [18445011909@trunkinbound:1] AGI("SIP/didforsale_in2-00000011", "agi-DID_route.agi") in new stack
[Dec 22 09:17:56]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Dec 22 09:17:56]     -- <SIP/didforsale_in2-00000011>AGI Script agi-DID_route.agi completed, returning 0
[Dec 22 09:17:56]     -- Executing [99909*2***DID@default:1] Answer("SIP/didforsale_in2-00000011", "") in new stack
[Dec 22 09:17:56]        > 0x7f1234051340 -- Strict RTP switching to RTP target address 64.125.111.109:26592 as source
[Dec 22 09:17:56]     -- Executing [99909*2***DID@default:2] AGI("SIP/didforsale_in2-00000011", "agi-VDAD_ALL_inbound.agi") in new stack
[Dec 22 09:17:56]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Dec 22 09:17:56]     -- <SIP/didforsale_in2-00000011> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:17:56]     -- <SIP/didforsale_in2-00000011> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:17:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 22 09:17:57]     -- Called 068*168*096*145*302@default
[Dec 22 09:17:57]     -- Executing [068*168*096*145*302@default:1] Goto("Local/068*168*096*145*302@default-00000009;2", "default,302,1") in new stack
[Dec 22 09:17:57]     -- Goto (default,302,1)
[Dec 22 09:17:57]     -- Executing [302@default:1] Dial("Local/068*168*096*145*302@default-00000009;2", "SIP/302,60,") in new stack
[Dec 22 09:17:57]   == Using SIP RTP CoS mark 5
[Dec 22 09:17:57]     -- Called SIP/302
[Dec 22 09:17:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 22 09:17:57]     -- Called 068*168*096*145*301@default
[Dec 22 09:17:57]     -- Executing [068*168*096*145*301@default:1] Goto("Local/068*168*096*145*301@default-0000000a;2", "default,301,1") in new stack
[Dec 22 09:17:57]     -- Goto (default,301,1)
[Dec 22 09:17:57]     -- Executing [301@default:1] Dial("Local/068*168*096*145*301@default-0000000a;2", "SIP/301,60,") in new stack
[Dec 22 09:17:57]   == Using SIP RTP CoS mark 5
[Dec 22 09:17:57]     -- Called SIP/301
[Dec 22 09:17:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 22 09:17:57]     -- Called 068*168*096*145*303@default
[Dec 22 09:17:57]     -- Executing [068*168*096*145*303@default:1] Goto("Local/068*168*096*145*303@default-0000000b;2", "default,303,1") in new stack
[Dec 22 09:17:57]     -- Goto (default,303,1)
[Dec 22 09:17:57]     -- Executing [303@default:1] Dial("Local/068*168*096*145*303@default-0000000b;2", "SIP/303,60,") in new stack
[Dec 22 09:17:57]   == Using SIP RTP CoS mark 5
[Dec 22 09:17:57]     -- Called SIP/303
[Dec 22 09:17:57]     -- SIP/301-00000013 is ringing
[Dec 22 09:17:57]     -- Local/068*168*096*145*301@default-0000000a;1 is ringing
[b][Dec 22 09:17:57]     -- SIP/301-00000013 is ringing
[Dec 22 09:17:57]     -- SIP/303-00000014 is ringing
[Dec 22 09:17:57]     -- SIP/303-00000014 is ringing[/b]
[Dec 22 09:17:57]     -- Local/068*168*096*145*303@default-0000000b;1 is ringing
[Dec 22 09:17:58]     -- SIP/302-00000012 is ringing
[Dec 22 09:17:58]     -- Local/068*168*096*145*302@default-00000009;1 is ringing
[Dec 22 09:17:58]     -- Started music on hold, class 'default', on channel 'SIP/didforsale_in2-00000011'
[Dec 22 09:18:01]        > 0x7f1234051340 -- Strict RTP learning complete - Locking on source address 64.125.111.109:26592
[Dec 22 09:18:01]     -- Stopped music on hold on SIP/didforsale_in2-00000011
[Dec 22 09:18:01]     -- <SIP/didforsale_in2-00000011> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:18:01]     -- <SIP/didforsale_in2-00000011> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:18:01]     -- <SIP/didforsale_in2-00000011> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:18:01]     -- <SIP/didforsale_in2-00000011> Playing 'generic_hold.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Dec 22 09:18:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 22 09:18:02]   == Manager 'sendcron' logged on from 127.0.0.1
[Dec 22 09:18:02]   == Manager 'sendcron' logged off from 127.0.0.1
[Dec 22 09:18:04]        > 0x7f11e4010390 -- Strict RTP learning after remote address set to: 192.168.0.134:64088
[b][Dec 22 09:18:04]     -- SIP/303-00000014 answered Local/068*168*096*145*303@default-0000000b;2[/b]
[Dec 22 09:18:04]     -- Local/068*168*096*145*303@default-0000000b;1 answered
[Dec 22 09:18:04]     -- Executing [138331*28330*Y2220917560000057510*303*@default:1] AGI("Local/068*168*096*145*303@default-0000000b;1", "agi-VDAD_local_optimize.agi,") in new stack
[Dec 22 09:18:04]     -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_local_optimize.agi
[Dec 22 09:18:04]     -- Channel SIP/303-00000014 joined 'simple_bridge' basic-bridge <91ac5e09-d057-4b08-ada5-c34eee5015e3>
[Dec 22 09:18:04]     -- Channel Local/068*168*096*145*303@default-0000000b;2 joined 'simple_bridge' basic-bridge <91ac5e09-d057-4b08-ada5-c34eee5015e3>
[Dec 22 09:18:04]     -- <Local/068*168*096*145*303@default-0000000b;1>AGI Script agi-VDAD_local_optimize.agi completed, returning 0
[Dec 22 09:18:04]     -- Executing [138331*28330*Y2220917560000057510*303*@default:2] Wait("Local/068*168*096*145*303@default-0000000b;1", "2") in new stack
[Dec 22 09:18:04]        > 0x7f11e4010390 -- Strict RTP qualifying stream type: audio
[Dec 22 09:18:04]        > 0x7f11e4010390 -- Strict RTP switching source address to 202.74.244.238:64088
[Dec 22 09:18:05]   == Manager 'sendcron' logged off from 127.0.0.1

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method (7.X.X?) and vicidial version with build (VERSION: 2.X-XXXx ... BUILD: #####-####).

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "manual/from scratch" you must post your operating system with version (and the .iso version from which you installed your original operating system) plus a link to the installation instructions you used. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Your example does not show the hangup.

4) Check SIP debug during the moment of hangup and the moments leading up to it. You can set the SIP debug to the IP of the phone you intend to answer to see if the handshake between the phone and the Vicidial server is the problem, then change it to the IP of the VoIP company to see if the problem is in the connection to the Carrier instead. One of those may provide a reason for the termination of the call.

5) Also ensure your Asterisk Version is correct in /etc/astguiclient.conf and admin->servers for this server. If it's wrong in either place you may need to re-install the sample config files for asterisk by performing a perl install.pl in the trunk folder. Note that the version in astguiclient.conf is tricky, some versions require an "X" in the version number. (EG: 11.X, not 11) This information is availalble in the install.pl script while it runs, it offers the versions and if it offers "11.X" and not "11", then you have to enter "11.X" for it to match properly at runtime for the installation of the proper sample scripts.