Can't establish call between 2 extensions

[dzigi]

Hi there,

I register carrier with account entry:

disallow=all
allow=alaw
type=friend
host=ip address
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very

and Dialplan entry:

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/CablenetSIP/${EXTEN:2},,tTor)
exten => _X.,3,Hangup

I create also 2 extensions 201 and 202 and try to make call.Bellow is what I got...

[Aug 30 14:34:07] -- Executing [202@default:1] Dial("SIP/201-00000006", "SIP/202|60|") in new stack
[Aug 30 14:34:07] WARNING[8171]: pbx.c:1442 pbx_exec: The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/202|60|))
[Aug 30 14:34:07] == Using SIP RTP CoS mark 5
[Aug 30 14:34:11] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 30 14:34:40] ERROR[8171]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("202|60|", "(null)", ...): System error
[Aug 30 14:34:40] WARNING[8171]: chan_sip.c:5865 create_addr: No such host: 202|60|
[Aug 30 14:34:40] WARNING[8171]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Aug 30 14:34:40] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 30 14:34:40] -- Executing [202@default:2] Goto("SIP/201-00000006", "default,85026666666666202,1") in new stack
[Aug 30 14:34:40] -- Goto (default,85026666666666202,1)
[Aug 30 14:34:40] -- Executing [85026666666666202@default:1] Wait("SIP/201-00000006", "1") in new stack
[Aug 30 14:34:41] -- Executing [85026666666666202@default:2] VoiceMail("SIP/201-00000006", "202,u") in new stack
[Aug 30 14:34:42] WARNING[8171]: file.c:666 ast_openstream_full: File vm-theperson does not exist in any format
[Aug 30 14:34:42] WARNING[8171]: file.c:957 ast_streamfile: Unable to open vm-theperson (format 0x4 (ulaw)): No such file or directory
[Aug 30 14:34:42] == Spawn extension (default, 85026666666666202, 2) exited non-zero on 'SIP/201-00000006'
[Aug 30 14:34:42] -- Executing [h@default:1] DeadAGI("SIP/201-00000006", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Aug 30 14:34:42] WARNING[8171]: res_agi.c:3930 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases!
[Aug 30 14:34:42] -- <SIP/201-00000006>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
linux-4mta*CLI>


Any idea what I done wrong.From SIP provider looks OK:

linux-4mta*CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
201/201 192.168.1.20 D N 51370 OK (101 ms)
202/202 192.168.1.124 D N 5060 OK (1 ms)
CablenetSIP 172.16.80.100 N 5060 OK (15 ms)

Thank you,

Ivan

Vicibox 5.0 from .iso | Vicidial VERSION: 2.8-408a BUILD: 130711-2208 | Asterisk 1.8.23.0| Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel |

[DomeDan]

The application delimiter is now the comma, not the pipe. Did you forget to convert your dialplan? (Dial(SIP/202|60|))

try to change it to comma in the dialplan and see if that helps

[dzigi]

Hi,I found problem between versions of Asterisk 1.4 on web administration and 1.8 if I check on CLI.So,I follow instructions from forum and I went to ADMIN->Server and I change Asterisk version to 1.8-vici and Rebuild Conf Files to Yes.I restart server and looks OK now-I establish internally call.

Now I am trying to make call outside and I am hearing ringback but other part doesn't hear me.I am sending log file when I make call...

[Aug 30 17:13:15] -- Executing [99862463@default:1] AGI("SIP/202-0000000f", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 30 17:13:15] -- <SIP/202-0000000f>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 30 17:13:15] -- Executing [99862463@default:2] Dial("SIP/202-0000000f", "SIP/CablenetSIP/862463,,tTor") in new stack
[Aug 30 17:13:15] == Using SIP RTP CoS mark 5
[Aug 30 17:13:15] -- Called SIP/CablenetSIP/862463
[Aug 30 17:13:16] -- SIP/CablenetSIP-00000010 is making progress passing it to SIP/202-0000000f
[Aug 30 17:13:33] == Spawn extension (default, 99862463, 2) exited non-zero on 'SIP/202-0000000f'
[Aug 30 17:13:33] -- Executing [h@default:1] DeadAGI("SIP/202-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
[Aug 30 17:13:33] WARNING[10100]: res_agi.c:3930 deadagi_exec: DeadAGI has been deprecated, please use AGI in all cases!
[Aug 30 17:13:33] -- <SIP/202-0000000f>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


I see that there is DeadAgi message(I had this also internally) maybe this is the problem,

thank you,

Ivan

[DomeDan]

Does not seam like you followed the instructions

The changes to Vicidial have just been committed for Asterisk 1.8 support. This is in the BETA testing phase currently, and is not recommended for use in production at this time. While we have performed a lot of testing, we have not tested this in production, so you have been warned.

If you choose to upgrade an existing system, please make note of the many conf file changes(extensions.conf and manager.conf) that will need to be made for Vicidial to function properly while interfacing with Asterisk 1.8. Also, make sure that the Admin -> Servers -> modify server "Asterisk Version" setting is set properly for the version you have installed on that server.

To download a version of Asterisk 1.8 that we are working from, go to:
http://downloads.vicidial.com/beta-apps/

To download the Vicidial code changes, just upgrade or download the latest svn/trunk snapshot.

"please make note of the many conf file changes(extensions.conf and manager.conf) that will need to be made"

That means that you need to copy the 1.8 config files from docs/conf_examples/ and apply any changes you wish afterwards