Calls dropping after 5 seconds

[dlilly31]

First post, so if i need more information please let me know. So we just got our carriers configured correctly. When we recieve an inbound, the call is connected and we can hear audio for about 5 seconds and then the call drops. Below is the debug report.

vicitel1*CLI>
[Mar 25 18:48:56] == Using SIP RTP CoS mark 5
[Mar 25 18:48:56] -- Executing [9499543503@trunkinbound:1] AGI("SIP/vitelity-in-00000009", "agi-DID_route.agi") in new stack
[Mar 25 18:48:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Mar 25 18:48:56] -- <SIP/vitelity-in-00000009>AGI Script agi-DID_route.agi completed, returning 0
[Mar 25 18:48:56] -- Executing [99909*2***DID@default:1] Answer("SIP/vitelity-in-00000009", "") in new stack
[Mar 25 18:48:56] -- Executing [99909*2***DID@default:2] AGI("SIP/vitelity-in-00000009", "agi-VDAD_ALL_inbound.agi") in new stack
[Mar 25 18:48:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Mar 25 18:48:56] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:56] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:56] > 0x7fcb0001d0a0 -- Probation passed - setting RTP source address to 66.241.99.236:18560
[Mar 25 18:48:57] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 25 18:48:57] -- Executing [010*210*000*076*78600051@default:1] Goto("Local/010*210*000*076*78600051@default-00000001;2", "default,78600051,1") in new stack
[Mar 25 18:48:57] -- Goto (default,78600051,1)
[Mar 25 18:48:57] -- Executing [78600051@default:1] MeetMe("Local/010*210*000*076*78600051@default-00000001;2", "8600051,Fq") in new stack
[Mar 25 18:48:57] > Channel Local/010*210*000*076*78600051@default-00000001;1 was answered
[Mar 25 18:48:57] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/010*210*000*076*78600051@default-00000001;1", "") in new stack
[Mar 25 18:48:57] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/010*210*000*076*78600051@default-00000001;1", "ding") in new stack
[Mar 25 18:48:57] -- <Local/010*210*000*076*78600051@default-00000001;1> Playing 'ding.gsm' (language 'en')
[Mar 25 18:48:57] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/010*210*000*076*78600051@default-00000001;1", "") in new stack
[Mar 25 18:48:57] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/010*210*000*076*78600051@default-00000001;1'
[Mar 25 18:48:57] -- Executing [h@vicidial-auto:1] AGI("Local/010*210*000*076*78600051@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Mar 25 18:48:57] -- <Local/010*210*000*076*78600051@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 25 18:48:57] == Spawn extension (default, 78600051, 1) exited non-zero on 'Local/010*210*000*076*78600051@default-00000001;2'
[Mar 25 18:48:57] -- Executing [h@default:1] AGI("Local/010*210*000*076*78600051@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Mar 25 18:48:57] -- <Local/010*210*000*076*78600051@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Mar 25 18:48:57] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:57] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:58] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 25 18:48:58] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:58] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:58] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:58] -- <SIP/vitelity-in-00000009> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'en')
[Mar 25 18:48:58] -- <SIP/vitelity-in-00000009>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Mar 25 18:48:58] -- Executing [010*210*000*076*8600051@default:1] Goto("SIP/vitelity-in-00000009", "default,8600051,1") in new stack
[Mar 25 18:48:58] -- Goto (default,8600051,1)
[Mar 25 18:48:58] -- Executing [8600051@default:1] MeetMe("SIP/vitelity-in-00000009", "8600051,F") in new stack
[Mar 25 18:49:01] == Manager 'sendcron' logged on from 127.0.0.1
[Mar 25 18:49:01] == Manager 'sendcron' logged off from 127.0.0.1
[Mar 25 18:49:02] WARNING[29579]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission 4ee7e75856993f3d51e657bd7f304c1c@66.241.99.236 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6401ms with no response
[Mar 25 18:49:02] WARNING[29579]: chan_sip.c:4067 retrans_pkt: Hanging up call 4ee7e75856993f3d51e657bd7f304c1c@66.241.99.236 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/ ... nsmissions).
[Mar 25 18:49:02] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/vitelity-in-00000009'
[Mar 25 18:49:02] -- Executing [h@default:1] AGI("SIP/vitelity-in-00000009", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----18---------------") in new stack
[Mar 25 18:49:02] -- <SIP/vitelity-in-00000009>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[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 and vicidial version with 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 "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. 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) I'm gonna go out on a limb here ... and say this may be related to your issue:

Mar 25 18:49:02] WARNING[29579]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission 4ee7e75856993f3d51e657bd7f304c1c@66.241.99.236 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/ ... nsmissions
Packet timed out after 6401ms with no response

do you have externip set to your public IP in /etc/asterisk/sip.conf?

otherwise, it could be a firewall issue (turn off the firewall to test this theory)

4) Happy Hunting!

[iboam]

I Have the same issue i just did a fresh install to my asterisk server (I Have a Cluster), my externip = public ip, even i tried with a different server (Express) same carrier and inbound settings and everything is fine. the issue is only on the cluster.