Need help configuring DialPlan with Vicidial

[Mark_Robinson]

Hi all,
Unsure of how to proceed with this, but here goes.
I have inherited support for our Vicidial servers whilst mnid way through changing carriers.
We have swapped the carrier to Gamma
What I have in the carrier details on Vicidial are as follows:

Account Entry
[Gamma]
type=peer
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
dtmfmode=rfc2833
context=default
canreinvite=no
allow=ulaw
allow=alaw
allow=g729
host=88.xxx.xxx.xxx


Dialplan Entry
exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944.,2,Dial(SIP/${EXTEN:1}@Gamma,,tTor)
exten => _944.,3,Hangup

CLI output when trying to make an outbound call:

[2020-05-14 08:12:32] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 08:12:32] Found
[2020-05-14 08:12:32] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 08:12:32] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-1c1b,2", "8600051|F") in new stack
[2020-05-14 08:12:32] > Channel Local/8600051@default-1c1b,1 was answered.
[2020-05-14 08:12:32] -- Executing [88447545704814@default:1] Answer("Local/8600051@default-1c1b,1", "") in new stack
[2020-05-14 08:12:32] -- Executing [88447545704814@default:2] PlayTones("Local/8600051@default-1c1b,1", "ring") in new stack
[2020-05-14 08:12:32] -- Executing [88447545704814@default:3] Dial("Local/8600051@default-1c1b,1", "IAX2/ASTloop:Mpe_123@127.0.0.1:40569/9447545704814||Tor") in new stack
[2020-05-14 08:12:32] -- Called ASTloop:Mpe_123@127.0.0.1:40569/9447545704814
[2020-05-14 08:12:32] -- Accepting AUTHENTICATED call from 127.0.0.1:
> requested format = gsm,
> requested prefs = (gsm|ulaw),
> actual format = ulaw,
> host prefs = (alaw|ulaw),
> priority = mine
[2020-05-14 08:12:32] -- Executing [9447545704814@default:1] AGI("IAX2/ASTloop-15907", "agi://127.0.0.1:4577/call_log") in new stack
[2020-05-14 08:12:32] -- Call accepted by 127.0.0.1 (format ulaw)
[2020-05-14 08:12:32] -- Format for call is ulaw
[2020-05-14 08:12:32] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TECH))
[2020-05-14 08:12:32] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2020-05-14 08:12:32] -- Executing [9447545704814@default:2] Dial("IAX2/ASTloop-15907", "SIP/447545704814@Gamma||tTo") in new stack
[2020-05-14 08:12:32] -- Called 447545704814@Gamma
[2020-05-14 08:12:32] -- SIP/Gamma-0000002e is circuit-busy
[2020-05-14 08:12:32] == Everyone is busy/congested at this time (1:0/1/0)
[2020-05-14 08:12:32] -- Executing [9447545704814@default:3] Hangup("IAX2/ASTloop-15907", "") in new stack
[2020-05-14 08:12:32] == Spawn extension (default, 9447545704814, 3) exited non-zero on 'IAX2/ASTloop-15907'
[2020-05-14 08:12:32] -- Executing [h@default:1] DeadAGI("IAX2/ASTloop-15907", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[2020-05-14 08:12:32] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 08:12:32] -- Hungup 'IAX2/ASTloop-15907'
[2020-05-14 08:12:32] -- Hungup 'IAX2/127.0.0.1:40569-13552'
[2020-05-14 08:12:32] == Everyone is busy/congested at this time (1:0/0/1)
[2020-05-14 08:12:32] -- Executing [88447545704814@default:4] Hangup("Local/8600051@default-1c1b,1", "") in new stack
[2020-05-14 08:12:32] == Spawn extension (default, 88447545704814, 4) exited non-zero on 'Local/8600051@default-1c1b,1'
[2020-05-14 08:12:32] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1c1b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CHANUNAVAIL----------") in new stack
[2020-05-14 08:12:32] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 08:12:33] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 08:12:33] Found
[2020-05-14 08:12:33] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 08:12:33] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-7973,2", "8600051|Fmq") in new stack
[2020-05-14 08:12:33] > Channel Local/58600051@default-7973,1 was answered.
[2020-05-14 08:12:33] -- Executing [8309@default:1] Answer("Local/58600051@default-7973,1", "") in new stack
[2020-05-14 08:12:33] -- Executing [8309@default:2] Monitor("Local/58600051@default-7973,1", "wav|TECH_20200514-081232_7545704814_906_2908067") in new stack
[2020-05-14 08:12:33] -- Executing [8309@default:3] Wait("Local/58600051@default-7973,1", "10800") in new stack
[2020-05-14 08:12:34] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-1c1b,2'
[2020-05-14 08:12:34] -- Executing [h@default:1] DeadAGI("Local/8600051@default-1c1b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[2020-05-14 08:12:34] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 08:12:34] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 08:12:34] Found



Please be gentle as i have no history of supporting this system at all.

[Kabis]

Hi

Did you check your carrier is registered in Asterisk CLI?

type

Code: Select all

asterisk -vvvvvvvvvvvvvvr

in console.

In that type

Code: Select all

sip show peers

.

Check the carrier status is okay or not?

If okay , we can proceed further. Else still your carrier is not registered.

[Mark_Robinson]

Hi Kabis,
Yes it shows as it is registered
Gamma 88.215.xxx.xxx 5060 OK (16 ms)

[Kabis]

Hi,

In Dialplan, You added 3 digits as prefix. But you eliminated 1 digit only(EXTEN:1).

So it eliminates 9 only. So is that right Dialplan?

Regards
KABIS

[Mark_Robinson]

Hi Kabis,
This is where i am getting confused, as i am a novice with Asterisk.

The numbers i am attempting to call are all in the UK
What do i need to put into my dial plan to dial the number 01302333222? (international would be +441302333222 (this is not the actual number i want to dial, just an example of the format & length)
In the campaign i have a Dial Prefix of 9 and a nothing in the Manual Dial Prefix

Thank you

[Kabis]

You are in UK. Or some other countries?

[Mark_Robinson]

I am based in the UK

[Kabis]

Then the SIP only make calls using Country code. So you use country code. Did you check the VOIP provider guidelines for call? Did they ask to put country code. And please add + and check. Or one time check without country code.

[Mark_Robinson]

i have changed the dial plan - still getting the following:

[2020-05-14 15:04:44] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-e2f4,2", "8600051|F") in new stack
[2020-05-14 15:04:44] > Channel Local/8600051@default-e2f4,1 was answered.
[2020-05-14 15:04:44] -- Executing [9447xxxxxxxxx@default:1] AGI("Local/8600051@default-e2f4,1", "agi://127.0.0.1:4577/call_log") in new stack
[2020-05-14 15:04:44] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=TECH))
[2020-05-14 15:04:44] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[2020-05-14 15:04:44] -- Executing [9447xxxxxxxxx@default:2] NoCDR("Local/8600051@default-e2f4,1", "") in new stack
[2020-05-14 15:04:44] -- Executing [9447xxxxxxxxx@default:3] Dial("Local/8600051@default-e2f4,1", "SIP/447xxxxxxxxx@Gamma||tTo") in new stack
[2020-05-14 15:04:44] -- Called 447xxxxxxxxx@Gamma
[2020-05-14 15:04:44] -- SIP/Gamma-00000048 is circuit-busy
[2020-05-14 15:04:44] == Everyone is busy/congested at this time (1:0/1/0)
[2020-05-14 15:04:44] -- Executing [9447xxxxxxxxx@default:4] Hangup("Local/8600051@default-e2f4,1", "") in new stack
[2020-05-14 15:04:44] == Spawn extension (default, 9447xxxxxxxxx, 4) exited non-zero on 'Local/8600051@default-e2f4,1'
[2020-05-14 15:04:44] -- Executing [h@default:1] DeadAGI("Local/8600051@default-e2f4,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----21-----CONGESTION----------") in new stack
[2020-05-14 15:04:44] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 15:04:45] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 15:04:45] Found
[2020-05-14 15:04:45] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 15:04:45] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-50c0,2", "8600051|Fmq") in new stack
[2020-05-14 15:04:45] > Channel Local/58600051@default-50c0,1 was answered.
[2020-05-14 15:04:45] -- Executing [8309@default:1] Answer("Local/58600051@default-50c0,1", "") in new stack
[2020-05-14 15:04:45] -- Executing [8309@default:2] Monitor("Local/58600051@default-50c0,1", "wav|TECH_20200514-150444_7xxxxxxxxx_906_2908209") in new stack
[2020-05-14 15:04:45] -- Executing [8309@default:3] Wait("Local/58600051@default-50c0,1", "10800") in new stack
[2020-05-14 15:04:45] == Manager 'sendcron' logged off from 127.0.0.1
[2020-05-14 15:04:46] == Manager 'sendcron' logged off from 127.0.0.1
[2020-05-14 15:04:47] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 15:04:47] Found
[2020-05-14 15:04:47] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 15:04:47] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-e2f4,2'
[2020-05-14 15:04:47] -- Executing [h@default:1] DeadAGI("Local/8600051@default-e2f4,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[2020-05-14 15:04:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 15:04:47] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 15:04:47] Found
[2020-05-14 15:04:47] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 15:04:47] == Parsing '/etc/asterisk/manager.conf': [2020-05-14 15:04:47] Found
[2020-05-14 15:04:47] == Manager 'sendcron' logged on from 127.0.0.1
[2020-05-14 15:04:47] == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-50c0,2'
[2020-05-14 15:04:47] -- Executing [h@default:1] DeadAGI("Local/58600051@default-50c0,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[2020-05-14 15:04:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[2020-05-14 15:04:47] == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-50c0,1'
[2020-05-14 15:04:47] -- Executing [h@default:1] DeadAGI("Local/58600051@default-50c0,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[2020-05-14 15:04:47] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --NODEBUG-

[Kabis]

Hi,

What did you change in Dialplan?

And your voip trunk not accepting this number. You need to check the dialplan for your voip provider. Are they accepting 10 digit with 44 prefix?
If yes, then check your credit balance in your voip provider also.

[williamconley]

Many carriers require the "International Long Distance" prefix of 011 before the country code. This differentiates from "domestic long distance" of "1" before the 10 digit NANPA number.

Code: Select all

exten=>_44XXXXXXXXXX,1,NoOp(UK)
exten=>_44XXXXXXXXXX,n,AGI(agi://127.0.0.1:4577/call_log)
exten=>_44XXXXXXXXXX,n,Dial(${CARRIER}/011${EXTEN},,To)
exten=>_44XXXXXXXXXX,n,Hangup

But this is carrier specific. They usually provide a help page with the expected prefix for international numbers along with a list of countries into which you may dial.

[Mark_Robinson]

Hi William,
I have sorted quite a lot of the problem out, however i need to add sip via header into the dial plan, and i have absolutely no idea of how to do that.
When we make a call, the local ip address is being sent in the header & the provider requires our LAN ip address to be sent instead.

[carpenox]

set your externip to your public ip address

in server settings: External Server IP:

[Mark_Robinson]

Hi Carpenox,I have set the External Server IP in the filed, however it still sends the internal one.

CLI output: [2020-05-15 13:49:27] Reliably Transmitting (no NAT) to 88.xxx.xxx.xxx:5060:
OPTIONS sip:88.xxx.xxx.xxx;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.xxx:5060;branch=z9hG4bK6ed868a4;rport
From: "asterisk" <sip:asterisk@192.168.xxx.xxx>;tag=as60c0bc9e
To: <sip:88.xxx.xxx.xxx;cpd=on>
Contact: <sip:asterisk@192.168.xxx.xxx>
Call-ID: 31273751143e8cfe67bfeac7787880a9@192.168.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 May 2020 12:49:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

[Kabis]

In /etc/asterisk/sip.conf,
externip=you set your public IP address

[Mark_Robinson]

Hi Kabis, I have set that also. Still the same error as above

[carpenox]

maybe try running the server ip script:

/usr/share/astguiclient/ADMIN_update_server_ip.pl

[williamconley]

in sip.conf is your local network included in one of the "localnet=" settings.

[Mark_Robinson]

in sip.conf is your local network included in one of the "localnet=" settings.

Hi William,
the local network is in localnet setting

As far as i can see, i have everything in place, but the call refuses to connect.

[Mark_Robinson]

maybe try running the server ip script:

/usr/share/astguiclient/ADMIN_update_server_ip.pl

I don't want to update the lan ip of the server.

[williamconley]

nat=yes for this sip account?

externip set to the public IP (on the other side of the nat) for the server?

restarted asterisk after these settings?

[Mark_Robinson]

nat=yes for this sip account?

externip set to the public IP (on the other side of the nat) for the server?

restarted asterisk after these settings?

Yeah done all that mate

[Mark_Robinson]

The provider says they need the following:


SIP Message
INVITE sip:01412485779@88.215.61.195 SIP/2.0
Via: SIP/2.0/UDP 88.215.25.5:5080;branch=z9hG4bK-a8b33135
From: "+441414040025" <sip:+441414040025@88.215.61.195>;tag=9fa8f5c74acc1965o0
To: <sip:01412485779@88.215.61.195>
Call-ID: 9899ceef-2d38d25d@192.168.1.8
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "+441414040025" sip:+441414040025@88.215.25.5:5080
Expires: 240
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 250
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp

and so far i have got:

OPTIONS sip:cc322@192.168.xxx.xxx:5060;rinstance=7fd83c3b13e0edf9;transport=UDP;cpd=on SIP/2.0
Via: SIP/2.0/UDP 88.xxx.xxx.xxx:5060;branch=z9hG4bK73d99fd6;rport
From: "asterisk" <sip:asterisk@185.xxx.xxx.xxx>;tag=as62cced3c
To: <sip:cc322@192.168.xxx.xxx:5060;rinstance=7fd83c3b13e0edf9;transport=UDP;cpd=on>
Contact: <sip:asterisk@88.xxx.xxx.xxx>
Call-ID: 10b5eef84e04e4551e612c84445e44e4@185.xxx.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 21 May 2020 11:28:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

192.168.xxx.xxx - lan ip address
185.xxx.xxx.xxx WAN ip address
85.xxx.xxx.xxx provider ip address.

How do i configure an INVITE header instead of an OPTIONS?

[williamconley]

Invite is "making a call". Options is "just checking with my server". This is not a configuration issue, the options packet is NOT a phone call. No phone call was generated.