Phones direct dial extensions not working

[ZeeTech]

Environment Details:
Scratch Install: OS CentOS 6.10
VERSION: 2.14-679a (BUILD: 180618-2300) VERSION: 2.14-679a: 180618-2300 | Asterisk 1.8.23.0-vici | Dell R730 | No additional HW

Problem: Dialer is working fine, agents login at the campaign and can receive and place call but internal dialing is not working.
For testing I setup two new phone accounts both phones are part of default context, when i dial 108 from 100 it hit to voicemail however second phone is connected.

CLI OutPut:
localhost*CLI> dialplan show 100@default
[ Included context 'vicidial-auto-phones' created by 'pbx_config' ]
'100' => 1. Dial(SIP/100|60|) [pbx_config]
2. Goto(default,85026666666666100,1) [pbx_config]
-= 1 extension (2 priorities) in 1 context. =-
localhost*CLI> dialplan show 108@default
[ Included context 'vicidial-auto-phones' created by 'pbx_config' ]
'108' => 1. Dial(SIP/108|60|) [pbx_config]
2. Goto(default,85026666666666108,1) [pbx_config]
-= 1 extension (2 priorities) in 1 context. =-

[Dec 10 15:26:20] == Using SIP RTP CoS mark 5
[Dec 10 15:26:20] -- Executing [108@default:1] Dial("SIP/100-0000003e", "SIP/108|60|") in new stack
[Dec 10 15:26:20] WARNING[28821]: pbx.c:1442 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/108|60|))
[Dec 10 15:26:20] == Using SIP RTP CoS mark 5
[Dec 10 15:26:21] ERROR[28821]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("108|60|", "(null)", ...): Name or service not known
[Dec 10 15:26:21] WARNING[28821]: chan_sip.c:5865 create_addr: No such host: 108|60|
[Dec 10 15:26:21] WARNING[28821]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Dec 10 15:26:21] == Everyone is busy/congested at this time (1:0/0/1)
[Dec 10 15:26:21] -- Executing [108@default:2] Goto("SIP/100-0000003e", "default,85026666666666108,1") in new stack
[Dec 10 15:26:21] -- Goto (default,85026666666666108,1)
[Dec 10 15:26:21] -- Executing [85026666666666108@default:1] Wait("SIP/100-0000003e", "1") in new stack
[Dec 10 15:26:22] -- Executing [85026666666666108@default:2] VoiceMail("SIP/100-0000003e", "108,u") in new stack
[Dec 10 15:26:22] -- <SIP/100-0000003e> Playing 'vm-theperson.gsm' (language 'en')
[Dec 10 15:26:24] -- <SIP/100-0000003e> Playing 'digits/1.gsm' (language 'en')
[Dec 10 15:26:24] -- <SIP/100-0000003e> Playing 'digits/0.gsm' (language 'en')
[Dec 10 15:26:25] -- <SIP/100-0000003e> Playing 'digits/8.gsm' (language 'en')
[Dec 10 15:26:26] -- <SIP/100-0000003e> Playing 'vm-isunavail.gsm' (language 'en')
[Dec 10 15:26:27] -- <SIP/100-0000003e> Playing 'vm-intro.gsm' (language 'en')
[Dec 10 15:26:27] == Spawn extension (default, 85026666666666108, 2) exited non-zero on 'SIP/100-0000003e'
[Dec 10 15:26:27] -- Executing [h@default:1] AGI("SIP/100-0000003e", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Dec 10 15:26:27] -- <SIP/100-0000003e>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
localhost*CLI>
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
100/100 10.20.30.192 D N 10104 OK (8 ms)
108/108 10.20.30.181 D N 5100 OK (4 ms)

[ZeeTech]

Fixed! Asterisk ver was incorrect in GUI under admin - > servers -> Asterisk Version

To fix click- > admin - > servers
1: Replace asterisk version 1.8.23.0 it was 1.4.- Sumbit
2. Rebuild conf files: Y - sumbit
3. Reboot Server

[williamconley]

Good postback.

Yep, you have to verify all information in admin->servers when you bring an old database to a new server. You should also review admin->system settings while you're in there.