Dialplan Entry

[Khanyasi]

We are using Vicidial version 2.6-381a with Asterisk 1.4.44-VICI, I am trying to set up my dialplan entry so we can call out from Canada to Irland and Trinidad, here I have the country and area codes
Irland 353-287-xxx-xxxx
Trinidad 868-788-xxxx

Here I have my Dailplan that i have made up to call out but no luck can someone please guide me correct dialplan please

exten => _35NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _86NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound2)

[blackbird2306]

What is your dial prefix setting in the campaign (default is 9) ? There are some problems with your dialplan.
First of all, where is your important call log line:
exten => _35NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
and then your 2nd line should be:
exten => _35NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)

Helpful:
https://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns

On the basis of default prefix "9" and not necessary to dial "011" (US exit code) in conjunction with your voip provider your dialplan could look like this (no guarantee ):
For calling to Trinidad there should be also the country code prefix "1" which is missing here (+1868) ?

exten => _9353NXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91868NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9353NXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _91868NXXXXXX,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9353NXXXXXXXXX,3,Hangup
exten => _91868NXXXXXX,3,Hangup

[Khanyasi]

Thanks for Help Blackbird here is the call_log line
[Nov 1 13:20:09] -- Executing [011353087721xxxx@defaultlog:1] AGI("SIP/298-00017dc7", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:20:09] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:20:09] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101132009_298_011353087721xxxx)
[Nov 1 13:20:09] -- <SIP/298-00017dc7>AGI Script agi-NVA_recording.agi completed,returning 0
[Nov 1 13:20:09] -- Executing [011353087721xxxx@defaultlog:2] Goto("SIP/298-00017dc7", "default,011353087721xxxx,1") in new stack
[Nov 1 13:20:09] -- Goto (default,011353087721xxxx,1)
[Nov 1 13:20:09] -- Channel 'SIP/298-00017dc7' sent to invalid extension: context,exten,priority=default,011353087721xxxx,1
[Nov 1 13:20:09] -- Executing [i@default:1] Playback("SIP/298-00017dc7", "invalid") in new stack
[Nov 1 13:20:09] > 0x7f4ae821a010 -- Probation passed - setting RTP source address to 192.168.2.105:49304
[Nov 1 13:20:09] -- <SIP/298-00017dc7> Playing 'invalid.gsm' (language 'en')
[Nov 1 13:20:13] == Spawn extension (default, i, 1) exited non-zero on 'SIP/298-00017dc7'
[Nov 1 13:20:13] -- Executing [h@default:1] AGI("SIP/298-00017dc7", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 1 13:20:13] -- <SIP/298-00017dc7>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
vici-brant*CLI>
Disconnected from Asterisk server
[Nov 1 13:20:14] Asterisk cleanly ending (0).
[Nov 1 13:20:14] Executing last minute cleanups


Also this is another entry in our dial plan if this helps

exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@nexco-outbound,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@modulis-outbound,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@modulis-outbound2,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound,,To)
exten => _1NXXNXXXXXX,n,Hangup

exten => s,n,Dial(SIP/${ARG1},5,rL(${call-limit}:call-limit-600))

exten => a,1,VoicemailMain(${EXTEN}@default)

Thanks

[Khanyasi]

Here i have just tested your suggestion but still no luck (last 4 digits of phone number being modified to xxxx)

[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:1] AGI("SIP/298-00017de5", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Nov 1 13:48:07] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Nov 1 13:48:07] -- AGI Script Executing Application: (Monitor) Options: (wav,/var/spool/asterisk/monitor/MIX/20171101134807_298_1868788xxxx)
[Nov 1 13:48:07] -- <SIP/298-00017de5>AGI Script agi-NVA_recording.agi completed, returning 0
[Nov 1 13:48:07] -- Executing [1868788xxxx@defaultlog:2] Goto("SIP/298-00017de5", "default,1868788xxxx,1") in new stack
[Nov 1 13:48:07] -- Goto (default,1868788xxxx,1)
[Nov 1 13:48:07] -- Executing [1868788xxxx@default:1] Dial("SIP/298-00017de5", "SIP/1868788xxxx@nexco-outbound,,To") in new stack
[Nov 1 13:48:07] == Using SIP RTP CoS mark 5
[Nov 1 13:48:07] -- Called SIP/1868788xxxx@nexco-outbound
[Nov 1 13:48:07] -- SIP/nexco-outbound-00017de6 answered SIP/298-00017de5
[Nov 1 13:48:07] > 0x7f4ae821a010 -- Probation passed - setting RTP source address to 192.168.2.105:57068
[Nov 1 13:48:07] > 0x7f4ae80ca890 -- Probation passed - setting RTP source address to 65.39.174.166:15516
[Nov 1 13:48:37] -- Executing [h@default:1] AGI("SIP/298-00017de5", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----30-----30") in new stack
[Nov 1 13:48:37] -- <SIP/298-00017de5>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -30-----30 completed, returning 0
[Nov 1 13:48:37] == Spawn extension (default, 1868788xxxx, 1) exited non-zero on 'SIP/298-00017de5'

[blackbird2306]

Please try it with "011" exit code and dial prefix "9" in campaign setting and after that without exit code.

Just for testing purpose use this one (but only for testing purposes because its not safe to allow all extensions, e.g. fraud, hack):

with campaign dial prefix "9":
exten => _9.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9.,2,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _9.,3,Hangup

or without prefix or "X" in campaign setting:
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/${EXTEN}@modulis-outbound2)
exten => _X.,3,Hangup

dial something like: "011 353 087721xxxx" or other combinations only "353 087721xxxx" and so on.
Now look if there is a connection and send us your asterisk CLI log.

Further tell us your vicidial calling scenario. Is it outbound calling with manual dialing, auto dialing or only from the softphone?

[Khanyasi]

I have tried your both suggestion but still can't make it here i have used the

or without prefix or "X" in campaign setting:
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/${EXTEN}@modulis-outbound2)
exten => _X.,3,Hangup


[Nov 2 15:49:25] -- Executing [71868788xxxx@default:1] Dial("SIP/298-0001837e", "SIP/1868788xxxx@modulis-outboun d") in new stack
[Nov 2 15:49:25] == Using SIP RTP CoS mark 5
[Nov 2 15:49:25] -- Called SIP/1868788xxxx@modulis-outbound
[Nov 2 15:49:25] -- SIP/modulis-outbound-0001837f answered SIP/298-0001837e
[Nov 2 15:49:25] -- Locally bridging SIP/298-0001837e and SIP/modulis-outbound-0001837f
[Nov 2 15:49:25] > 0x7f4ae8081313 -- Probation passed - setting RTP source address to 192.168.2.105:63602
[Nov 2 15:49:25] > 0x7f4b50060780 -- Probation passed - setting RTP source address to 199.182.133.203:18794
[Nov 2 15:49:32] -- Executing [h@default:1] AGI("SIP/516-0001837a", "agi://127.0.0.1:4577/call_log--HVcauses--PR I-----NODEBUG-----16-----ANSWER-----49-----28") in new stack
[Nov 2 15:49:32] -- <SIP/516-0001837a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... BUG-----16 -----ANSWER-----49-----28 completed, returning 0
[Nov 2 15:49:32] == Spawn extension (default, 19028754156, 1) exited non-zero on 'SIP/516-0001837a'
[Nov 2 15:49:35] -- Executing [h@default:1] AGI("SIP/298-0001837e", "agi://127.0.0.1:4577/call_log--HVcauses--PR I-----NODEBUG-----16-----ANSWER-----10-----10") in new stack
[Nov 2 15:49:35] -- <SIP/298-0001837e>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... BUG-----16 -----ANSWER-----10-----10 completed, returning 0
[Nov 2 15:49:35] == Spawn extension (default, 71868788xxxx, 1) exited non-zero on 'SIP/298-0001837e'

[blackbird2306]

We need much more information. There are many weird and paradoxal things:
Executing [71868788xxxx@default:1] Dial("SIP/298-0001837e", "SIP/1868788xxxx@modulis-outboun d") but using
exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/${EXTEN}@modulis-outbound2)

Please provide us with:
1. Admin --> carriers --> for every carrier in your listing ( Carrier ID, Registration String, Account Entry, Protocol, Globals String, Dialplan Entry)
2. Campaign settings: Dial Prefix, Manual Dial Prefix and Dial method
3. Please talk to your carrier provider(s) how the dialing format must be to connect to Ireland and Trinidad or whether its possible to call these routes at all

[Khanyasi]

Sure thanks, i will get in touch with carriers as well, here is requested information
Carrier ID: Modulis
Reg string: N/A

Account entry:
[modulis-outbound]
host=gw05.modulis.ca
type=friend
dtmfmode=rfc2833
canreinvite=no
insecure=port,invite
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw

Protocol: SIP
Global String: N/A

Dial Entry:
e911 must be enabled. see DIDs > NPANXXNXXX > Action menu > e911

exten => _71NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound)
exten => _718XXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound)

exten => _41NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound2)
exten => _418XXNXXXXXX,1,Dial(SIP/${EXTEN:1}@modulis-outbound2)



exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@nexco-outbound,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@modulis-outbound,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@modulis-outbound2,,To)
exten => _1NXXNXXXXXX,n,Dial(SIP/${EXTEN}@vitel-outbound,,To)
exten => _1NXXNXXXXXX,n,Hangup

exten => s,n,Dial(SIP/${ARG1},5,rL(${call-limit}:call-limit-600))

exten => a,1,VoicemailMain(${EXTEN}@default)

ring camb office

exten => 200,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 200,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 200,n,Dial(${CAMPHONES},15,Ttr)
exten => 200,n,Wait,2
exten => 200,n,Playback(/var/spool/asterisk/voicemail/default/281/unavail)
exten => 200,n,VoiceMail(281@default)

ring Bran office all exten

exten => 500,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 500,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 500,n,Dial(SIP/501&SIP/502&SIP/503&SIP/504&SIP/505&SIP/506&SIP/507&SIP/508&SIP/509&SIP/510&SIP/511&SIP/513&SIP/514&SIP/516&SIP/517,30,Ttr)
exten => 500,n,Wait,2
exten => 500,n,Playback(/var/spool/asterisk/voicemail/default/502/unavail)
exten => 500,n,VoiceMail(502@default)


exten => 210,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 210,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 210,n,Dial(SIP/282&SIP/294,30,Ttr)
exten => 210,n,Wait,2
exten => 210,n,Playback(/var/spool/asterisk/voicemail/default/282/unavail)
exten => 210,n,VoiceMail(282@default)

exten => 800,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 800,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 800,n,Dial(SIP/801&SIP/802&SIP/803&SIP/804&SIP/805&SIP/806,30,Ttr)
exten => 800,n,Wait,2
exten => 800,n,Playback(/var/spool/asterisk/voicemail/default/801/unavail)
exten => 800,n,VoiceMail(801@default)

Camb

exten => 3202,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 3202,n,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => 3202,n,Wait,2
exten => 3202,n,Playback(/var/lib/asterisk/sounds/44048020)
exten => 3202,n,Wait,2
exten => 3202,n,Playback(/var/lib/asterisk/sounds/85100075)
exten => 3202,n,Wait,2
exten => 3202,n,Dial(SIP/202,15,Ttr)
exten => 3202,n,Wait,2
exten => 3202,n,Playback(/var/lib/asterisk/sounds/85100031)
exten => 3202,n,Dial(SIP/202&SIP/511&SIP/285&SIP/805,15,Ttr)
exten => 3202,n,Wait,2
exten => 3202,n,Playback(/var/spool/asterisk/voicemail/default/202/unavail)
exten => 3202,n,VoiceMail(202@default)

[blackbird2306]

Some questions:
1. Who set this diaplan up? First of all there is a problem with missing "call_log" line as mentioned above.
First line of dialplan must be "exten => xxxx,1,AGI(agi://127.0.0.1:4577/call_log)" and then "exten => xxxx,2,Dial..." ! This is essential to make vicidial work properly. It's not an option.

2. Is this vicidial installation already in production for some time? Did you ever make successfull calls with these settings within Canada? If yes, please send us asterisk output for these calls (does it still work?).

3. You send us only account entry for "modulis-outbound". Where is the rest "modulis-outbound2", "nexco-outbound" and "vitel-outbound". And which one do you want to use for outbound calls to Trinidad or Ireland?

3. How do you use vicidial? How do you initiate the calls (agent mask or softphone). Inbound calls or outbound calls?

4. Send us all your campaign settings. Each little thing in the campaign settings can make things go wrong (missing character is enough)

5. We need asterisk debug output for the faulty calls.
From Command line:
1. start the asterisk console: "asterisk -vvvvvvvvvvvvvvvvvvvvr"
the next few steps should be done quickly without activity on the system or you will have a lot output
2. "sip set debug on"
3. make a call
4. "sip set debug off"
5. "exit" (if you want to close asterisk cli)

[Khanyasi]

Thanks for your help I was using the wrong carrier at first but finally it worked I have just used the entry shown below in Dialplan Entry under our Nexco SIP provider and it worked like a charm.
exten => _011353.,1,Dial(SIP/${EXTEN}@nexco-outbound)