Hi,
I am having a problem with my DID configuration. Outbound dialing is fine, it's Inbound that's not working correctly.
Here's the problem. I tried to call my DID using my other sip account. It connects then suddenly ends.
Here is my Carrier Settings
[SIP1010]
disallow=all
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=invite
nat=yes
host=sg.sip.commpeak.com
username=XXXXXXXX
secret=XXXXXXXX
allow=alaw
Here's my Entry
exten => _6920542949.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _6920542949.,2,Dial(SIP/${EXTEN:10}@SIP1010,,tTo)
exten => _6920542949.,3,Hangup
I created a DID pointing to my IN_GROUP and that is allowed and being used on my Campaign. Once I logged in to agent interface, I select that Inbound setting. But when I try to call the DID, I get this log
== Using SIP RTP CoS mark 5
-- Executing [18189620828@trunkinbound:1] AGI("SIP/SIP1010-000000b1", "agi-DID_route.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
-- <SIP/SIP1010-000000b1>AGI Script agi-DID_route.agi completed, returning 0
-- Executing [99909*1***DID@default:1] Answer("SIP/SIP1010-000000b1", "") in new stack
-- Executing [99909*1***DID@default:2] AGI("SIP/SIP1010-000000b1", "agi-VDAD_ALL_inbound.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
-- <SIP/SIP1010-000000b1>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
-- Executing [99909*1***DID@default:3] Hangup("SIP/SIP1010-000000b1", "") in new stack
== Spawn extension (default, 99909*1***DID, 3) exited non-zero on 'SIP/SIP1010-000000b1'
-- Executing [h@default:1] AGI("SIP/SIP1010-000000b1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- <SIP/SIP1010-000000b1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
It connects to my server then the call suddenly ends.
I'm using GoAutodial Asterisk 1.8.23.0-1_centos7.go RPM by demian@goautodial.com built by root @ centos7.goautodial.com on a x86_64 running Linux on 2014-08-01 00:18:09 UTC
Server Specs
2vCPU 4GBRAM
CentOS7
Appreciate all the help!