[Solved]Dialer does not want to talk to voip provider

[davesdatasystems]

So this is a huge issue at the moment, and i am hoping someone here might know something i have not tried yet.

Xcast is the voip provider, Verizon Fios is the internet provider.

I am trying to see if you guys might think this is Verizon Fios or the router some how blocking me from talking to the voip provider.

First my dial plan, i have used this dial plan forever and never had a issue

Newest Version of the preloaded VICIDIAL btw. However all other stats are the same that are listed in my signature.
Vicibox_v.6.0.x86_64-6.0.3.preload.iso

[Xcast]
host=38.102.250.50
type=friend
disallow=all
allow=ulaw
trustrpid=yes
sendrpid=yes
qualify=yes
insecure=port,invite
dtmfmode=rfc2833
canreinvite=no

exten=> _81XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten=> _81XXXXXXXXXX,2,Dial(SIP/Xcast/${EXTEN:1},,tTr)
exten=> _81XXXXXXXXXX,3,Hangup

CLI Spam
When i try to make a outgoing call in predictive

[Jul 2 14:41:58] NOTICE[5541]: chan_local.c:973 local_call: No such extension/context 812037922708@default while calling Local channel
[Jul 2 14:41:58] NOTICE[5541]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922708@default
[Jul 2 14:41:58] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:58] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:58] NOTICE[5545]: chan_local.c:973 local_call: No such extension/context 812037922741@default while calling Local channel
[Jul 2 14:41:58] NOTICE[5545]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922741@default
[Jul 2 14:41:58] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:58] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:58] NOTICE[5548]: chan_local.c:973 local_call: No such extension/context 812037922745@default while calling Local channel
[Jul 2 14:41:58] NOTICE[5548]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922745@default
[Jul 2 14:41:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:59] NOTICE[5551]: chan_local.c:973 local_call: No such extension/context 812037922828@default while calling Local channel
[Jul 2 14:41:59] NOTICE[5551]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922828@default
[Jul 2 14:41:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:59] NOTICE[5554]: chan_local.c:973 local_call: No such extension/context 812037922920@default while calling Local channel
[Jul 2 14:41:59] NOTICE[5554]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922920@default
[Jul 2 14:41:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:59] NOTICE[5558]: chan_local.c:973 local_call: No such extension/context 812037922941@default while calling Local channel
[Jul 2 14:41:59] NOTICE[5558]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922941@default
[Jul 2 14:41:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:59] NOTICE[5561]: chan_local.c:973 local_call: No such extension/context 812037922945@default while calling Local channel
[Jul 2 14:41:59] NOTICE[5561]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922945@default
[Jul 2 14:41:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:59] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:41:59] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:41:59] NOTICE[5564]: chan_local.c:973 local_call: No such extension/context 812037922956@default while calling Local channel
[Jul 2 14:41:59] NOTICE[5564]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922956@default
[Jul 2 14:42:00] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:42:00] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:42:00] NOTICE[5568]: chan_local.c:973 local_call: No such extension/context 812037922958@default while calling Local channel
[Jul 2 14:42:00] NOTICE[5568]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922958@default
[Jul 2 14:42:00] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:42:00] == Manager 'sendcron' logged off from 127.0.0.1


When i try to make a outgoing call manual
== Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:24] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000017;2", "8600051,F") in new stack
[Jul 2 14:45:24] > Channel Local/8600051@default-00000017;1 was answered.
[Jul 2 14:45:24] == Starting Local/8600051@default-00000017;1 at default,817272710085,1 failed so falling back to exten 's'
[Jul 2 14:45:24] -- Executing [s@default:1] wait("Local/8600051@default-00000017;1", "1")
[Jul 2 14:45:24] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:24] NOTICE[6512]: chan_local.c:973 local_call: No such extension/context 812037922987@default while calling Local channel
[Jul 2 14:45:24] NOTICE[6512]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037922987@default
[Jul 2 14:45:25] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:25] NOTICE[6515]: chan_local.c:973 local_call: No such extension/context 812037923093@default while calling Local channel
[Jul 2 14:45:25] NOTICE[6515]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923093@default
[Jul 2 14:45:25] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:25] NOTICE[6518]: chan_local.c:973 local_call: No such extension/context 812037923157@default while calling Local channel
[Jul 2 14:45:25] NOTICE[6518]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923157@default
[Jul 2 14:45:25] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:25] -- Executing [s@default:1] answer("Local/8600051@default-00000017;1", "")
[Jul 2 14:45:25] -- Digit timeout set to 5.000
[Jul 2 14:45:25] -- Response timeout set to 10.000
[Jul 2 14:45:25] -- Executing [s@default:1] background("Local/8600051@default-00000017;1", "demo-congrats")
[Jul 2 14:45:25] -- <Local/8600051@default-00000017;1> Playing 'demo-congrats.gsm' (language 'en')
[Jul 2 14:45:25] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:25] NOTICE[6522]: chan_local.c:973 local_call: No such extension/context 812037923181@default while calling Local channel
[Jul 2 14:45:25] NOTICE[6522]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923181@default
[Jul 2 14:45:25] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:25] NOTICE[6526]: chan_local.c:973 local_call: No such extension/context 812037923233@default while calling Local channel
[Jul 2 14:45:25] NOTICE[6526]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923233@default
[Jul 2 14:45:25] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:25] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:25] NOTICE[6529]: chan_local.c:973 local_call: No such extension/context 812037923247@default while calling Local channel
[Jul 2 14:45:25] NOTICE[6529]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923247@default
[Jul 2 14:45:26] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:26] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:26] NOTICE[6533]: chan_local.c:973 local_call: No such extension/context 812037923349@default while calling Local channel
[Jul 2 14:45:26] NOTICE[6533]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923349@default
[Jul 2 14:45:26] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:26] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:26] NOTICE[6536]: chan_local.c:973 local_call: No such extension/context 812037923390@default while calling Local channel
[Jul 2 14:45:26] NOTICE[6536]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923390@default
[Jul 2 14:45:26] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:26] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:26] NOTICE[6539]: chan_local.c:973 local_call: No such extension/context 812037923415@default while calling Local channel
[Jul 2 14:45:26] NOTICE[6539]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923415@default
[Jul 2 14:45:26] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:26] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:26] NOTICE[6542]: chan_local.c:973 local_call: No such extension/context 812037923431@default while calling Local channel
[Jul 2 14:45:26] NOTICE[6542]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923431@default
[Jul 2 14:45:26] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:26] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:26] NOTICE[6546]: chan_local.c:973 local_call: No such extension/context 812037923451@default while calling Local channel
[Jul 2 14:45:26] NOTICE[6546]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923451@default
[Jul 2 14:45:27] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:27] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:27] NOTICE[6549]: chan_local.c:973 local_call: No such extension/context 812037923455@default while calling Local channel
[Jul 2 14:45:27] NOTICE[6549]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923455@default
[Jul 2 14:45:27] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:27] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:27] NOTICE[6552]: chan_local.c:973 local_call: No such extension/context 812037923603@default while calling Local channel
[Jul 2 14:45:27] NOTICE[6552]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923603@default
[Jul 2 14:45:27] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:27] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:27] NOTICE[6556]: chan_local.c:973 local_call: No such extension/context 812037923636@default while calling Local channel
[Jul 2 14:45:27] NOTICE[6556]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923636@default
[Jul 2 14:45:27] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:27] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:27] NOTICE[6559]: chan_local.c:973 local_call: No such extension/context 812037923673@default while calling Local channel
[Jul 2 14:45:27] NOTICE[6559]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923673@default
[Jul 2 14:45:27] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:27] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:27] NOTICE[6562]: chan_local.c:973 local_call: No such extension/context 812037923692@default while calling Local channel
[Jul 2 14:45:27] NOTICE[6562]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923692@default
[Jul 2 14:45:28] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:28] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:28] NOTICE[6565]: chan_local.c:973 local_call: No such extension/context 812037923760@default while calling Local channel
[Jul 2 14:45:28] NOTICE[6565]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923760@default
[Jul 2 14:45:28] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:28] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:28] NOTICE[6569]: chan_local.c:973 local_call: No such extension/context 812037923771@default while calling Local channel
[Jul 2 14:45:28] NOTICE[6569]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923771@default
[Jul 2 14:45:28] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:28] NOTICE[6572]: chan_local.c:973 local_call: No such extension/context 812037923774@default while calling Local channel
[Jul 2 14:45:28] NOTICE[6572]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923774@default
[Jul 2 14:45:28] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:28] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:28] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:28] NOTICE[6575]: chan_local.c:973 local_call: No such extension/context 812037923869@default while calling Local channel
[Jul 2 14:45:28] NOTICE[6575]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923869@default
[Jul 2 14:45:28] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:28] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:28] NOTICE[6578]: chan_local.c:973 local_call: No such extension/context 812037923928@default while calling Local channel
[Jul 2 14:45:28] NOTICE[6578]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037923928@default
[Jul 2 14:45:29] == Manager 'sendcron' logged off from 127.0.0.1
[Jul 2 14:45:29] == Manager 'sendcron' logged on from 127.0.0.1
[Jul 2 14:45:29] NOTICE[6582]: chan_local.c:973 local_call: No such extension/context 812037924004@default while calling Local channel
[Jul 2 14:45:29] NOTICE[6582]: channel.c:5529 __ast_request_and_dial: Unable to call channel Local/812037924004@default


Ip is set correctly in the sip file, the router is configured to DMZ for right now for testing, so it should be opened to the world, and i have did the /usr/share/astguiclient/ADMIN_update_server_ip.pl
after the install.

[dransiksapoc]

I don't think that's an issue with your trunk provider, but verify, setup a softphone directly with the sip info and see if you can make outbound calls.

[davesdatasystems]

Not my trunk provider, i am pretty sure (unless there is something i am missing) i am more worried about it being my ISP

[dransiksapoc]

It's a simple first step, I mean. If your box is local, setup the softphone to eliminate it being an issue with asterisk or vici. If it works on the softphone then, there you go...

[davesdatasystems]

Softphone connects to dialer just fine. The dialer just wont talk to the voip provider

[davesdatasystems]

ERROR[2963]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("vicidial", "(null)", ...): Name or service not known
[Jul 2 17:22:02] WARNING[2963]: acl.c:590 resolve_first: Unable to lookup 'vicidial'


I am pretty sure this is where my issue is, But i cant seem to figure out how to get past it.

[dransiksapoc]

I'm not talking about the softphone connecting to the dialer I'm talking the softphone connecting directly to your trunk provider. That being said where are you getting that error? What are you running that's generating it? Restarting httpd?

[davesdatasystems]

It is in the asterisk CLI, it is happening every time i try to do something with the asterisk cli, i am working on a fix for that part right now. I have a simple zoiper softphone, not sure how to connect it directly to my voip provider (xcast)

[Acidshock]

Think you need a space between exten and the =>

[davesdatasystems]

Thank you for the help Acidshock, sadly it did not change anything at all and gave me all the same errors.

[Acidshock]

No problem. Have you checked to see if /etc/asterisk/extensions-vicidial.conf is updated with your new dialplan? Could be a problem where the config file isn't getting created/rebuilt.

[Acidshock]

This may also sound dumb but you do have the carrier entry set to Active? I have forgot to set this option more times than I like to admit.

Do you have the second part with your exten => _81XXXXXXXXXX in the second part of the carrier config called Dialplan Entry?

[davesdatasystems]

I guess i did not post, so if it did, this will be a delayed double post but allows me to answer the question you ask. Also sorry if this is annoying. I do thank you for the help.
Yes, the carrier is set to active (i have did that in the past as well) Yes the exten information is in the second box.

Also i am kind of clueless about the extensions-vicidial.conf so i am going to post it here and see if you might see a issue with it, something does not look right to me, but i could be way off the mark.

THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
TRUNKloop = IAX2/ASTloop:oFuN5Wt6kiMWcsr@127.0.0.1:40569
TRUNKblind = IAX2/ASTblind:oFuN5Wt6kiMWcsr@127.0.0.1:41569
TRUNKplay = IAX2/ASTplay:oFuN5Wt6kiMWcsr@127.0.0.1:42569




; agent phones restricted to only internal extensions
[default---agent]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----default---agent-------------------------NO)
exten => s,n,Set(INVCOUNT=0)
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)


; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; custom dialplan entries
include => vicidial-auto-internal
include => vicidial-auto-phones




; logging of all outbound calls from agent phones
[defaultlog]
exten => s,1,Answer
exten => s,n,AGI(agi-VDAD_inbound_calltime_check.agi,-----NO-----defaultlog-------------------------NO)
exten => s,n,Set(INVCOUNT=0)
exten => s,n,Background(sip-silence)
exten => s,n,WaitExten(20)


; hangup
exten => t,1,Playback(vm-goodbye)
exten => t,n,Hangup()
exten => i,1,Goto(s,4)
exten => i,n,Hangup()
; hangup
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; custom dialplan entries
exten => _X.,1,AGI(agi-NVA_recording.agi,BOTH------Y---Y---Y)
exten => _X.,n,Goto(default,${EXTEN},1)




[vicidial-auto-external]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Local Server: 192.168.1.200
exten => _192*168*001*200*.,1,Goto(default,${EXTEN:16},1)
exten => _192*168*001*200*.,2,Hangup()
exten => _**192*168*001*200*.,1,Goto(default,${EXTEN:18},1)
exten => _**192*168*001*200*.,2,Hangup()

; Agent session audio playback meetme entry
exten => _473782178600XXX,1,Meetme(${EXTEN:8},q)
exten => _473782178600XXX,n,Hangup()
; Agent session audio playback loop
exten => _473782168600XXX,1,Dial(${TRUNKplay}/47378217${EXTEN:8},5,To)
exten => _473782168600XXX,n,Hangup()
; Agent session audio playback extension
exten => 473782158521111,1,Answer
exten => 473782158521111,n,ControlPlayback(${CALLERID(name)},99999,0,1,2,3,4)
exten => 473782158521111,n,Hangup()
; SendDTMF to playback channel to control it
exten => _473782148521111.,1,Answer
exten => _473782148521111.,n,SendDTMF(${CALLERID(num)},250,250,IAX2/ASTplay-${EXTEN:15})
exten => _473782148521111.,n,Hangup()
; Silent wait channel for DTMFsend
exten => 473782138521111,1,Answer
exten => 473782138521111,n,Wait(5)
exten => 473782138521111,n,Hangup()
; VICIDIAL Carrier: Xcast - Xcast
; Xcast
exten => _81XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81XXXXXXXXXX,2,Dial(SIP/Xcast/${EXTEN:1},,tTr)
exten => _81XXXXXXXXXX,3,Hangup

[vicidial-auto-internal]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})
; Voicemail Extensions:
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,n,Voicemail(${EXTEN:14},u)
exten => _85026666666666.,n,Hangup()
exten => _85026666666667.,1,Wait(1)
exten => _85026666666667.,n,Voicemail(${EXTEN:14},su)
exten => _85026666666667.,n,Hangup()
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)
exten => 8500,3,Hangup()
exten => 8501,1,VoicemailMain(s${CALLERID(num)})
exten => 8501,2,Hangup()

; Prompt Extensions:
exten => 8167,1,Answer
exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000)
exten => 8167,3,Hangup()
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000)
exten => 8168,3,Hangup()

; this is used for recording conference calls, the client app sends the filename
; value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
; Recording is limited to 1 hour, to make longer, just change the server
; setting ViciDial Recording Limit
; this is the WAV verison, default
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERID(name)})
exten => 8309,3,Wait(3600)
exten => 8309,4,Hangup()
; this is the GSM verison
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERID(name)})
exten => 8310,3,Wait(3600)
exten => 8310,4,Hangup()

; agent alert extension
exten => 83047777777777,1,Answer
exten => 83047777777777,2,Playback(${CALLERID(name)})
exten => 83047777777777,3,Hangup()
; This is a loopback dial-around to allow for immediate answer of outbound calls
exten => _8305888888888888.,1,Answer
exten => _8305888888888888.,n,Wait(${EXTEN:16:1})
exten => _8305888888888888.,n,Dial(${TRUNKloop}/${EXTEN:17},,To)
exten => _8305888888888888.,n,Hangup()
; No-call silence extension
exten => _8305888888888888X999,1,Answer
exten => _8305888888888888X999,n,Wait(3600)
exten => _8305888888888888X999,n,Hangup()

[vicidial-auto-phones]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Phones direct dial extensions:
exten => 5001,1,Dial(SIP/5001,60,)
exten => 5001,2,Goto(default,850266666666665001,1)
exten => 5001,3,Hangup()
exten => 102,1,Dial(SIP/gs102,60,)
exten => 102,2,Goto(default,85026666666666102,1)
exten => 102,3,Hangup()

[vicidial-auto]
exten => h,1,AGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

include => vicidial-auto-internal
include => vicidial-auto-phones
include => vicidial-auto-external


; END OF FILE Last Forced System Reload: 2015-07-02 12:36:01

[Acidshock]

put context=trunkinbound in the account entry section at the bottom of [Xcast]

[Acidshock]

Keep in mind it takes a few minutes to rebuild and reload. I typically leave a terminal up with the asterisk console running to see when it reloads.

[davesdatasystems]

I guess when i copy and pasted i missed the last few entries, but context=trunkinbound is there


The full entry without missing anything is
[Xcast]
host=38.102.250.50
type=friend
disallow=all
allow=ulaw
trustrpid=yes
sendrpid=yes
qualify=yes
insecure=port,invite
dtmfmode=rfc2833
canreinvite=no
nat=no
host=38.102.250.60
context=trunkinbound

[davesdatasystems]

Keep in mind it takes a few minutes to rebuild and reload. I typically leave a terminal up with the asterisk console running to see when it reloads.

I give it plenty of time, if i dont completly restart it when i make a change. I have never had this issue before, well i have, but it was easy to resolve this one is kicking my butt.

[Acidshock]

Can you post your /etc/asterisk/sip.conf file? Also check your PMs. Be careful when posting the sip.conf not to include any passwords.

[davesdatasystems]

Removed the external ip address for safety reason. I am sure you understand, but it is set to my static external ip




[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = XX.XX.XX.XX ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000

[Acidshock]

You do have 2 host entries in the carrier settings. Not sure if that would invalidate it. If you type "sip show peers" in asterisk does Xcast show up?

[davesdatasystems]

Yeah, i tried to remove the second host, the second host is for inbound, but i had no such luck, everything stayed the same

[Acidshock]

We got it fixed. Seems like /etc/extensions.conf was missing the #include extensions-vicidial.conf

[davesdatasystems]

I want to thank Acidshock for the help, I hope one day i can return the favor

[williamconley]

We got it fixed. Seems like /etc/extensions.conf was missing the #include extensions-vicidial.conf

And good postback to boot.

Note that any updates/upgrades should always include loading the sample .conf files (during the install.pl phase) and extensions.conf should NEVER be edited directly. All changes should be in custom dialplan entries in the Vicidial GUI

[davesdatasystems]

Good advice, however i did not update it manually, I tried a new voip provider (via request of my client) and i think somehow it messed with that file. I never seen the issue before and never had this problem before. But i would like to tell people the voip provider that gave me issues was called Rubix.

[williamconley]

That file can not be modified in the Vicidial Web interface. It can only be modified by someone logging in to the console (ssh or directly) and making a change. If you gave the Telco access, they broke it. LOL