New Vicibox installed can't make internal calls

[Khanyasi]

Hello Experts, I have just installed new Vicidial v7.0.3 on an i7 12 GB Ram pc I have completed all steps in the installation including OS-install, network configuration, Firewall set to internal on one of the network card can ping internally as well as to ping outside, vicibox-express installation is done, I have created few user's and phones using vicidial admin.php interface, no sip carriers and dial plan being setup yet, I have configured all extension on different SNOM 710 phones so when I call local phone extension I get message the extension 291 is not available and it goes in to voicemail, can someone please help out with this situation.
below is Asterisk logs for dialing local extension.
Thanks,

[Feb 7 11:30:05] == Using SIP RTP CoS mark 5
[Feb 7 11:30:05] WARNING[1917][C-00000002]: chan_sip.c:33714 process_crypto: Ignoring crypto attribute in SDP because RTP transport is insecure
[Feb 7 11:30:05] -- Executing [291@defaultlog:1] AGI("SIP/298-00000002", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Feb 7 11:30:05] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Feb 7 11:30:05] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20180207113005_298_291)
[Feb 7 11:30:05] WARNING[4345][C-00000002]: pbx.c:1682 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Monitor(wav|/var/spool/asterisk/monitor/MIX/20180207113005_298_291))
[Feb 7 11:30:05] WARNING[4345][C-00000002]: file.c:1242 ast_writefile: No such format 'wav|/var/spool/asterisk/monitor/MIX/20180207113005_298_291'
[Feb 7 11:30:05] WARNING[4345][C-00000002]: res_monitor.c:363 __ast_monitor_start: Could not create file /var/spool/asterisk/monitor/audio-in-2
[Feb 7 11:30:05] WARNING[4345][C-00000002]: res_monitor.c:571 __ast_monitor_change_fname: Cannot change monitor filename of channel SIP/298-00000002 to null
[Feb 7 11:30:05] -- <SIP/298-00000002>AGI Script agi-NVA_recording.agi completed, returning 0
[Feb 7 11:30:05] -- Executing [291@defaultlog:2] Goto("SIP/298-00000002", "default,291,1") in new stack
[Feb 7 11:30:05] -- Goto (default,291,1)
[Feb 7 11:30:05] -- Executing [291@default:1] Dial("SIP/298-00000002", "SIP/291|60|") in new stack
[Feb 7 11:30:05] WARNING[4345][C-00000002]: pbx.c:1682 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/291|60|))
[Feb 7 11:30:05] == Using SIP RTP CoS mark 5
[Feb 7 11:30:05] ERROR[4345][C-00000002]: netsock2.c:271 ast_sockaddr_resolve: getaddrinfo("291|60|", "(null)", ...): Name or service not known
[Feb 7 11:30:05] WARNING[4345][C-00000002]: chan_sip.c:6108 create_addr: No such host: 291|60|
[Feb 7 11:30:05] WARNING[4345][C-00000002]: app_dial.c:2455 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Feb 7 11:30:05] == Everyone is busy/congested at this time (1:0/0/1)
[Feb 7 11:30:05] -- Executing [291@default:2] Goto("SIP/298-00000002", "default,85026666666666291,1") in new stack
[Feb 7 11:30:05] -- Goto (default,85026666666666291,1)
[Feb 7 11:30:05] -- Executing [85026666666666291@default:1] Wait("SIP/298-00000002", "1") in new stack
[Feb 7 11:30:06] == Manager 'sendcron' logged on from 127.0.0.1
[Feb 7 11:30:06] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 11:30:06] -- Executing [85026666666666291@default:2] VoiceMail("SIP/298-00000002", "291,u") in new stack
[Feb 7 11:30:07] > 0x7f787c010350 -- Probation passed - setting RTP source address to 192.168.2.105:60550
[Feb 7 11:30:07] -- <SIP/298-00000002> Playing 'vm-theperson.gsm' (language 'en')
[Feb 7 11:30:08] -- <SIP/298-00000002> Playing 'digits/2.gsm' (language 'en')
[Feb 7 11:30:09] -- <SIP/298-00000002> Playing 'digits/9.gsm' (language 'en')
[Feb 7 11:30:10] -- <SIP/298-00000002> Playing 'digits/1.gsm' (language 'en')
[Feb 7 11:30:10] -- <SIP/298-00000002> Playing 'vm-isunavail.gsm' (language 'en')
[Feb 7 11:30:11] == Manager 'sendcron' logged off from 127.0.0.1
[Feb 7 11:30:11] -- <SIP/298-00000002> Playing 'vm-intro.gsm' (language 'en')
[Feb 7 11:30:17] -- <SIP/298-00000002> Playing 'beep.gsm' (language 'en')
[Feb 7 11:30:17] -- Recording the message
[Feb 7 11:30:17] -- x=0, open writing: /var/spool/asterisk/voicemail/default/291/tmp/CGuGfo format: wav49, 0x7f78a8007128
[Feb 7 11:30:17] -- x=1, open writing: /var/spool/asterisk/voicemail/default/291/tmp/CGuGfo format: gsm, 0x7f78a8005b58
[Feb 7 11:30:17] -- x=2, open writing: /var/spool/asterisk/voicemail/default/291/tmp/CGuGfo format: wav, 0x7f78a8006318
[Feb 7 11:30:18] -- User hung up
[Feb 7 11:30:18] == Spawn extension (default, 85026666666666291, 2) exited non-zero on 'SIP/298-00000002'
[Feb 7 11:30:18] -- Executing [h@default:1] AGI("SIP/298-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Feb 7 11:30:18] -- <SIP/298-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
ViciFlorida*CLI>

[ambiorixg12]

Fix this and it will posibly work

b 7 11:30:05] -- Executing [291@default:1] Dial("SIP/298-00000002", "SIP/291|60|") in new stack
[Feb 7 11:30:05] WARNING[4345][C-00000002]: pbx.c:1682 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/291|60|))

[Khanyasi]

Sure, I will try this and let you know, Thank you.

[Khanyasi]

I did change that but it went back to the way it was after restart i used grep to find the string as shwon below than made changes to that path
Here is path where i have made this change but it revert back after reboot

viciflorida:/etc/asterisk # grep -r 'Dial(SIP/291|60|)' /etc/asterisk/
/etc/asterisk/extensions-vicidial.conf:exten => 291,1,Dial(SIP/291|60|)

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

; Phones direct dial extensions:
exten => 291,1,Dial(SIP/291|60|)
exten => 291,2,Goto(default,85026666666666291,1)
exten => 298,1,Dial(SIP/298|60|)
exten => 298,2,Goto(default,85026666666666298,1)
exten => 601,1,Dial(SIP/601|60|)
exten => 601,2,Goto(default,85026666666666601,1)
exten => 102,1,Dial(SIP/gs102|60|)
exten => 102,2,Goto(default,85026666666666102,1)

[vicidial-auto]
exten => h,1,DeadAGI(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: 2018-02-09 11:28:44
You have new mail in /var/mail/root

[Khanyasi]

Thanks ambiorixg12, I have found solution for my problem as shown below, i can make internal calls but asterisk cli is still showing pbx.c:1682 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan?

Re: Soundboard Error
Unread postby zenithbsolutions » Thu Jun 29, 2017 3:07 am

Hi,

Found the solution for updating the configs.

1 - Admin -> Servers
2 - Generate conf files: N
3 - Rebuild conf files: Y
4 - Rebuild Music On Hold: Y
5 - Sounds Update: Y

Hope it helps you. Thank you
zenithbsolutions

Posts: 61
Joined: Tue Aug 25, 2015 2:17 am

[Feb 13 09:15:31] Connected to Asterisk 11.25.3-vici currently running on viciflorida (pid = 2334)
[Feb 13 09:15:36] == Using SIP RTP CoS mark 5
[Feb 13 09:15:36] WARNING[2418][C-00000004]: chan_sip.c:33714 process_crypto: Ignoring crypto attribute in SDP because RTP transport is insecure
[Feb 13 09:15:36] > 0x7f39fc0009b0 -- Strict RTP learning after remote address set to: 192.168.2.105:51294
[Feb 13 09:15:36] -- Executing [601@defaultlog:1] AGI("SIP/298-00000007", "agi-NVA_recording.agi,BOTH------Y---Y---Y") in new stack
[Feb 13 09:15:36] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-NVA_recording.agi
[Feb 13 09:15:36] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20180213091536_298_601)
[Feb 13 09:15:36] WARNING[4324][C-00000004]: pbx.c:1682 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Monitor(wav|/var/spool/asterisk/monitor/MIX/20180213091536_298_601))
[Feb 13 09:15:36] WARNING[4324][C-00000004]: file.c:1242 ast_writefile: No such format 'wav|/var/spool/asterisk/monitor/MIX/20180213091536_298_601'
[Feb 13 09:15:36] WARNING[4324][C-00000004]: res_monitor.c:370 __ast_monitor_start: Could not create file /var/spool/asterisk/monitor/audio-in-4
[Feb 13 09:15:36] WARNING[4324][C-00000004]: res_monitor.c:578 __ast_monitor_change_fname: Cannot change monitor filename of channel SIP/298-00000007 to null
[Feb 13 09:15:36] -- <SIP/298-00000007>AGI Script agi-NVA_recording.agi completed, returning 0
[Feb 13 09:15:36] -- Executing [601@defaultlog:2] Goto("SIP/298-00000007", "default,601,1") in new stack
[Feb 13 09:15:36] -- Goto (default,601,1)
[Feb 13 09:15:36] -- Executing [601@default:1] Dial("SIP/298-00000007", "SIP/601,60,") in new stack
[Feb 13 09:15:36] == Using SIP RTP CoS mark 5
[Feb 13 09:15:36] -- Called SIP/601
[Feb 13 09:15:36] -- SIP/601-00000008 is ringing
[Feb 13 09:15:36] -- SIP/601-00000008 is ringing
[Feb 13 09:15:37] -- SIP/601-00000008 is ringing
[Feb 13 09:15:39] -- SIP/601-00000008 is ringing
[Feb 13 09:15:40] == Spawn extension (default, 601, 1) exited non-zero on 'SIP/298-00000007'
[Feb 13 09:15:40] -- Executing [h@default:1] AGI("SIP/298-00000007", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[Feb 13 09:15:40] -- <SIP/298-00000007>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
viciflorida*CLI>