Autodial calls are hung up. Manual calls work

[petr]

When calling in autodial-mode, the call is connected but is hung up after ca 3 seconds. Manual calls work fine. There is sound coming through when autodialing...

These are my carrier settings and dialplan:

[phonera]
disallow=all
allow=gsm
allow=ulaw
type=peer
dtmfmode=rfc2833
context=trunkoutbound
qualify=yes
insecure=very
nat=yes
host=HOST
username=USER
secret=PASSWORD

Dialplan
exten => _00X!,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _00X!,2,Dial(sip/${EXTEN}@SIPTRUNK,55,o)
exten => _00X!,3,Hangup

Asterisk -r output (replaced phonenr, ip etc):
[Jun 12 11:02:02] -- Executing [MYPHONENUMBER@default:1] AGI("Local/MYPHONENUMBER@default-58ad,2", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 12 11:02:02] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 12 11:02:02] -- Executing [MYPHONENUMBER@default:2] Dial("Local/MYPHONENUMBER@default-58ad,2", "sip/MYPHONENUMBER@SIPTRUNK|55|o") in new stack
[Jun 12 11:02:02] -- Called MYPHONENUMBER@SIPTRUNK
[Jun 12 11:02:07] == Parsing '/etc/asterisk/manager.conf': [Jun 12 11:02:07] Found
[Jun 12 11:02:07] == Manager 'sendcron' logged on from 192.168.1.1
[Jun 12 11:02:07] == Manager 'sendcron' logged off from 192.168.1.1
[Jun 12 11:02:10] -- SIP/SIPTRUNK-00000004 is ringing
[Jun 12 11:02:19] -- SIP/SIPTRUNK-00000004 answered Local/MYPHONENUMBER@default-58ad,2
[Jun 12 11:02:19] > Channel Local/MYPHONENUMBER@default-58ad,1 was answered.
[Jun 12 11:02:19] -- Executing [8368@default:1] Playback("Local/MYPHONENUMBER@default-58ad,1", "sip-silence") in new stack
[Jun 12 11:02:19] -- <Local/MYPHONENUMBER@default-58ad,1> Playing 'sip-silence' (language 'en')
[Jun 12 11:02:19] WARNING[5588]: file.c:1297 waitstream_core: Unexpected control subclass '-1'
[Jun 12 11:02:19] -- Executing [8368@default:2] AGI("Local/MYPHONENUMBER@default-58ad,1", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 12 11:02:19] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Jun 12 11:02:19] -- Executing [8368@default:3] AGI("Local/MYPHONENUMBER@default-58ad,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Jun 12 11:02:19] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jun 12 11:02:20] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jun 12 11:02:20] -- Executing [8368@default:4] AGI("Local/MYPHONENUMBER@default-58ad,1", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
[Jun 12 11:02:20] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Jun 12 11:02:21] == Manager 'sendcron' logged off from 192.168.1.1
[Jun 12 11:02:21] -- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
[Jun 12 11:02:21] -- Executing [8368@default:5] Hangup("Local/MYPHONENUMBER@default-58ad,1", "") in new stack
[Jun 12 11:02:21] == Spawn extension (default, 8368, 5) exited non-zero on 'Local/MYPHONENUMBER@default-58ad,1'
[Jun 12 11:02:21] -- Executing [h@default:1] DeadAGI("Local/MYPHONENUMBER@default-58ad,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Jun 12 11:02:22] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jun 12 11:02:22] -- Executing [h@default:1] DeadAGI("Local/MYPHONENUMBER@default-58ad,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----3") in new stack
[Jun 12 11:02:23] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --20-----3 completed, returning 0
[Jun 12 11:02:23] == Spawn extension (default, MYPHONENUMBER, 2) exited non-zero on 'Local/MYPHONENUMBER@default-58ad,2'

Any ideas?

Regards
Petr

[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 think there's a problem with your search and replace here, but in case this is real, please explain to me why "Playback" occurs here:

[Jun 12 11:02:19] > Channel Local/MYPHONENUMBER@default-58ad,1 was answered.
[Jun 12 11:02:19] -- Executing [8368@default:1] Playback("Local/MYPHONENUMBER@default-58ad,1", "sip-silence") in new stack

4) Given the odds you changed your IP address (and forgot to mention that?). Also, I note your partial information regarding "there is sound coming through when autodialing ...". I know this sounds clear to you from your perspective, since you're in the office and saw the results (and heard them). But this is not an indication of "Two Way Sound" which is necessary for a completed phone call. If two-way-sound is never established, asterisk will leave the call local (not answered, not initiating the rest of the call). If the call is still local, that means there is no two-way-sound and Vicidial will flat-out refuse to pass the call to an agent (waste of time, right? agent won't be able to talk and listen), so it terminates the call instead. If you meant to say "the called party can hear the survey recording", please say that next time instead of "there's sound coming through". One-way sound can be caused by "externip" being incorrect in sip.conf. It should be set to your external IP address. Also worthy of note, insecure should not be very any more, that's deprecated.

5) For future reference, when you have both a "working" and a "not working", it is customary to post both and tag them as such. Helps sometimes.

Happy Hunting!