vicidial.org

Hi Guys,

Good day to all! I got some problems which i cant figure out for 2 days already. I hope somebody can see it and help me point into the proper direction.

TIA

------------------------------
Problem calling inbound. Call cannot pass thru the extensions 10001 or 10002. If i register an extension no. the same as the authname. Inbound calls can be received. Only 1 extension work. How is that so? Did i miss something?


Linux 4.1.39-56-default
Asterisk 11.25.1-vici

register => authname:secret@ipaddressofcarrier:5060/authname

[carrier]
disallow=all
allow=ulaw
allow=gsm
allow=alaw
username=authname
fromuser=authname
type=friend
secret=secret
qualify=no
host=ipaddofcarrier
fromdomain=ipaddofcarrier
insecure=invite
dtmfmode=rfc2833
defaultexpirey=60
nat=no
canreinvite=no
context=trunkinbound

Protocal: SIP

Global String: SIPtrunk=SIP/carrier

Dialplan Entry:
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

Server IP: 192.168.0.1
Active: Y

Have you read the Vicidial Manager Manual?

How do you have the DID's routing configured in your Vicidial web admin?

Could you provide Asterisk CLI output of the call coming in?

Hi,

-- when using the authentication name as the extension this is the log

[Jul 12 03:10:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 03:10:38] == Using SIP RTP CoS mark 5
[Jul 12 03:10:38] -- Executing [967516516@default:1] Dial("SIP/Nextiva-00000010", "SIP/967516516,60,") in new stack
[Jul 12 03:10:38] == Using SIP RTP CoS mark 5
[Jul 12 03:10:38] -- Called SIP/967516516
[Jul 12 03:10:39] -- SIP/967516516-00000011 is ringing
[Jul 12 03:11:00] -- SIP/967516516-00000011 answered SIP/Nextiva-00000010
[Jul 12 03:11:00] -- Locally bridging SIP/Nextiva-00000010 and SIP/967516516-00000011
[Jul 12 03:11:00] > 0x7fe640044c80 -- Probation passed - setting RTP source address to 192.168.1.11:60630
[Jul 12 03:11:00] > 0x7fe64003a4b0 -- Probation passed - setting RTP source address to 208.73.146.95:17812
[Jul 12 03:11:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 03:11:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 03:11:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 03:11:02] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 03:11:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 03:11:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 03:11:07] -- Executing [h@default:1] AGI("SIP/Nextiva-00000010", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----29-----7") in new stack
[Jul 12 03:11:07] -- <SIP/Nextiva-00000010>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --29-----7 completed, returning 0
[Jul 12 03:11:07] == Spawn extension (default, 967516516, 1) exited non-zero on 'SIP/Nextiva-00000010'


-- when using another extension no. 1002 this is the log

[Jul 12 03:12:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 12 03:12:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 12 03:12:08] -- Unregistered SIP '967516516'
[Jul 12 03:12:08] -- Registered SIP '1002' at 192.168.1.11:54923
[Jul 12 03:12:08] -- Unregistered SIP '1002'
[Jul 12 03:12:08] -- Registered SIP '1002' at 192.168.1.11:54923
[Jul 12 03:12:16] == Using SIP RTP CoS mark 5
[Jul 12 03:12:16] -- Executing [967516516@default:1] Dial("SIP/Nextiva-00000012", "SIP/967516516,60,") in new stack
[Jul 12 03:12:16] WARNING[15943][C-0000000c]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Jul 12 03:12:16] == Everyone is busy/congested at this time (1:0/0/1)
[Jul 12 03:12:16] -- Executing [967516516@default:2] Goto("SIP/Nextiva-00000012", "default,85026666666666967516516,1") in new stack
[Jul 12 03:12:16] -- Goto (default,85026666666666967516516,1)
[Jul 12 03:12:16] -- Executing [85026666666666967516516@default:1] Wait("SIP/Nextiva-00000012", "1") in new stack
[Jul 12 03:12:17] -- Executing [85026666666666967516516@default:2] VoiceMail("SIP/Nextiva-00000012", "967516516,u") in new stack
[Jul 12 03:12:18] -- <SIP/Nextiva-00000012> Playing 'vm-theperson.gsm' (language 'en')
[Jul 12 03:12:18] > 0x7fe64004d4e0 -- Probation passed - setting RTP source address to 208.73.146.95:25706
[Jul 12 03:12:19] -- <SIP/Nextiva-00000012> Playing 'digits/9.gsm' (language 'en')
[Jul 12 03:12:20] -- <SIP/Nextiva-00000012> Playing 'digits/6.gsm' (language 'en')
[Jul 12 03:12:21] -- <SIP/Nextiva-00000012> Playing 'digits/7.gsm' (language 'en')
[Jul 12 03:12:22] -- <SIP/Nextiva-00000012> Playing 'digits/5.gsm' (language 'en')
[Jul 12 03:12:22] -- <SIP/Nextiva-00000012> Playing 'digits/1.gsm' (language 'en')
[Jul 12 03:12:23] -- <SIP/Nextiva-00000012> Playing 'digits/6.gsm' (language 'en')
[Jul 12 03:12:24] -- <SIP/Nextiva-00000012> Playing 'digits/5.gsm' (language 'en')
[Jul 12 03:12:24] -- <SIP/Nextiva-00000012> Playing 'digits/1.gsm' (language 'en')
[Jul 12 03:12:25] -- <SIP/Nextiva-00000012> Playing 'digits/6.gsm' (language 'en')
[Jul 12 03:12:26] -- <SIP/Nextiva-00000012> Playing 'vm-isunavail.gsm' (language 'en')
[Jul 12 03:12:27] -- <SIP/Nextiva-00000012> Playing 'vm-intro.gsm' (language 'en')



Thanks in advance.

These look like outbound calls, not inbound calls.

No problem with outbound calls, these are the output from the CLI if someone outside calls the DID.
extension 967516516 can receive while extension 1002 cannot receive any calls. it plays the audio recording... 967516516 is not available.

I don't see the calls coming into [trunkinbound] like they should be, so something appears to be misconfigured.

[Jul 12 03:12:16] -- Executing [967516516@default:1] Dial("SIP/Nextiva-00000012", "SIP/967516516,60,") in new stack
[Jul 12 03:12:16] WARNING[15943][C-0000000c]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

this goes out everytime I call the other extensions.

is there a separate dialplan setting for inbound calls?

In your inbound SIP carrier entry, putting the context=trunkinbound should get those inbound calls to go to that context.

Hi mflorell,

Thank you for your patience and taking time to respond to my questions. You will find below my carrier entry settings.

[Nextiva]
disallow=all
allow=ulaw
allow=gsm
allow=alaw
username=967XXXXXX
fromuser=967XXXXXX
type=friend
secret=secret
qualify=no
host=XXX.XX.XX.XX
fromdomain=XXX.XX.XX.XX
insecure=invite
dtmfmode=rfc2833
defaultexpirey=60
nat=no
canreinvite=no
context=trunkinbound

Have you tried placing the call into this system from a phone that is not on this system?

Yes, i used skype and another phone that is entirely not on my system.

i tried skype and another phone different from the system. The replies are the same. "It says the number you dialed is not yet in service."


[Jul 12 23:32:32] -- Executing [967516516@trunkinbound:1] AGI("SIP/Nextiva-00000001", "agi-DID_route.agi") in new stack
[Jul 12 23:32:32] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jul 12 23:32:32] -- <SIP/Nextiva-00000001>AGI Script agi-DID_route.agi completed, returning 0
[Jul 12 23:32:32] -- Executing [9998811112@default:1] Wait("SIP/Nextiva-00000001", "2") in new stack
[Jul 12 23:32:34] -- Executing [9998811112@default:2] Answer("SIP/Nextiva-00000001", "") in new stack
[Jul 12 23:32:35] -- Executing [9998811112@default:3] Playback("SIP/Nextiva-00000001", "ss-noservice") in new stack
[Jul 12 23:32:35] -- <SIP/Nextiva-00000001> Playing 'ss-noservice.gsm' (language 'en')
[Jul 12 23:32:40] -- Executing [9998811112@default:4] Playback("SIP/Nextiva-00000001", "vm-goodbye") in new stack
[Jul 12 23:32:40] -- <SIP/Nextiva-00000001> Playing 'vm-goodbye.gsm' (language 'en')
[Jul 12 23:32:41] == Spawn extension (default, 9998811112, 4) exited non-zero on 'SIP/Nextiva-00000001'
[Jul 12 23:32:41] -- Executing [h@default:1] AGI("SIP/Nextiva-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 12 23:32:41] -- <SIP/Nextiva-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jul 12 23:32:50] == Using SIP RTP CoS mark 5
[Jul 12 23:32:50] -- Executing [967516516@trunkinbound:1] AGI("SIP/Nextiva-00000002", "agi-DID_route.agi") in new stack
[Jul 12 23:32:50] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jul 12 23:32:50] -- <SIP/Nextiva-00000002>AGI Script agi-DID_route.agi completed, returning 0
[Jul 12 23:32:50] -- Executing [9998811112@default:1] Wait("SIP/Nextiva-00000002", "2") in new stack
[Jul 12 23:32:52] -- Executing [9998811112@default:2] Answer("SIP/Nextiva-00000002", "") in new stack
[Jul 12 23:32:52] -- Executing [9998811112@default:3] Playback("SIP/Nextiva-00000002", "ss-noservice") in new stack
[Jul 12 23:32:52] -- <SIP/Nextiva-00000002> Playing 'ss-noservice.gsm' (language 'en')
[Jul 12 23:32:56] == Spawn extension (default, 9998811112, 3) exited non-zero on 'SIP/Nextiva-00000002'
[Jul 12 23:32:56] -- Executing [h@default:1] AGI("SIP/Nextiva-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Jul 12 23:32:56] -- <SIP/Nextiva-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

Do you have a DID entry for "967516516" in your web config?

If so, what is it set to route to?

Hi mflorell,

I rechecked everything, luckily thru your help i can now be able to call and pass thru the DID to the extension. Now, i will try to pass it on to more extensions.
Thanks a lot! Your help is greatly appreciated.

Thanks for the post-back!