vicidial.org

Posted: **Sat May 12, 2012 1:42 am**

When I manually place a call out of x-lite, the calls do not go through. I don't get any error message in x-lite, just doesn't ring and the call never reaches its destination number. Any ideas?

Carrier is a voip provider, connected to via SIP.

AGI("SIP/100-00000030", "agi://127.0.0.1:4577/call_log") in new stack
[May 12 00:27:58] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[May 12 00:27:58] -- Executing [91780863****@default:2] Dial("SIP/100-00000030", "SIP/voipms/7808633745||tTor") in new stack
[May 12 00:27:58] -- Called voipms/780863****
[May 12 00:27:58] -- SIP/voipms-00000031 answered SIP/100-00000030
[May 12 00:28:01] == Parsing '/etc/asterisk/manager.conf': [May 12 00:28:01] Found
[May 12 00:28:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:28:01] == Parsing '/etc/asterisk/manager.conf': [May 12 00:28:01] Found
[May 12 00:28:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:28:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:28:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:28:06] == Parsing '/etc/asterisk/manager.conf': [May 12 00:28:06] Found
[May 12 00:28:06] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:28:06] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:28:10] -- Executing [h@default:1] DeadAGI("SIP/100-00000030", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----12") in new stack
[May 12 00:28:10] WARNING[10215]: res_agi.c:2212 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI
[May 12 00:28:10] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -12-----12 completed, returning 0
[May 12 00:28:10] == Spawn extension (default, 91780863****, 2) exited non-zero on 'SIP/100-00000030'
[May 12 00:28:18] WARNING[2805]: chan_sip.c:2058 retrans_pkt: Maximum retries exceeded on transmission YjM5ZTcxNTZhOTM3MGFkNDAxNDgxNmQwMmY1NGFkNzM. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
[May 12 00:29:01] == Parsing '/etc/asterisk/manager.conf': [May 12 00:29:01] Found
[May 12 00:29:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:29:01] == Parsing '/etc/asterisk/manager.conf': [May 12 00:29:01] Found
[May 12 00:29:01] == Manager 'sendcron' logged on from 127.0.0.1
[May 12 00:29:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:29:01] == Manager 'sendcron' logged off from 127.0.0.1
[May 12 00:29:06] == Parsing '/etc/asterisk/manager.conf': [May 12 00:29:06] Found

VERSION: 2.4-357a
BUILD: 120125-2107

Posted: **Sat May 12, 2012 1:45 am**

Here is my carrier settings too:

ACCOUNT
[voipms]
insecure=very
canreinvite=no
context=trunkinbound
host=64.120.**.**
secret=*****
type=friend
username=******
allow=ulaw
allow=alaw
trustrpid=yes
requirecalltoken=no

GLOBALS

TESTSIPTRUNK = SIP/voipms

DIALPLAN

exten => _91XXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXNXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91XXXNXXXXXX,3,Hangup

Posted: **Sun May 13, 2012 4:11 pm**

Hvw you checked in admin under users i am sure their is a setting in there to allow manual calls. Was going to ask if you had 91 at front but se you do,

Posted: **Mon May 14, 2012 7:36 pm**

I have this problem even dialing strait out of the softphone.

Posted: **Tue May 15, 2012 2:32 am**

Hi
I would check your carrier settings, if your server is behind a nat you'll need nat=yes in the carrier settings, I can't see it in your example.
Just follow what voipms recommends for a server behind nat and then retest.

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