Hi ,
My Problem is when I dial an extension within LAN there is a ring tone and phone does ring on the other side but after couple of seconds it goes to the voice mail. This happens, even when I click talk on the other phone it still goes to Voicemail. I am using x lite for this testing
My realtime setup looks fine and i have created an interface that can create the extension from a web site but Problem is that there is no communication
My Current Environment is asterisknow. Realtime
Asterisk 1.4.24, Copyright © 1999 - 2008 Digium, Inc. and others.
Linux version CENTOS 5.3
Data from Extensions table ;
| 9 | incoming | 5006 | 1 | Dial | SIP/5006 ,30|
| 10 | incoming | 5006 | 2 | Voicemail | u5006 |
| 11 | incoming | 5006 | 3 | Hangup | |
| 12 | incoming | 8500 | 1 | VoicemailMain | |
| 14 | incoming | 5007 | 1 | Dial | SIP/5007,30 |
| 15 | incoming | 5007 | 2 | Voicemail | u5007 |
| 16 | incoming | 5007 | 3| Hangup | |
—- ———- ——- ———- ————— ———-
In addition to that I have 1 Wired Router supplied by the ISP with one port and then I have another Dlink N wireless router which is being used as wireless access point and No router functionality is utilized for this router .I have opened RTP and 5060 Ports on my Router provided by ISP.
here is the log
== Spawn extension (incoming, 5008, 2) exited non-zero on ‘SIP/5009-09f52f38’
– Executing Dial(“SIP/5008-09f51918”, “SIP/5009”)
– Called 5009
== Spawn extension (incoming, 5009, 1) exited non-zero on ‘SIP/5008-09f51918’
– Executing Dial(“SIP/5008-09f51918”, “SIP/5009”)
– Called 5009
== Spawn extension (incoming, 5009, 1) exited non-zero on ‘SIP/5008-09f51918’
– Executing Dial(“SIP/5009-09f4cbd8”, “SIP/5008”)
– Called 5008
– SIP/5008-09f8e730 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Voicemail(“SIP/5009-09f4cbd8”, “u5008”)
– <SIP/5009-09f4cbd8> Playing ‘vm-theperson’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘digits/5’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘digits/0’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘digits/0’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘digits/8’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘vm-isunavail’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘vm-intro’ (language ‘en’)
– <SIP/5009-09f4cbd8> Playing ‘beep’ (language ‘en’)
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/5008/tmp/H0s0fm format: wav49, 0x9f5b2a0
– x=1, open writing: /var/spool/asterisk/voicemail/default/5008/tmp/H0s0fm format: wav, 0x9f4f260
– User hung up
– Recording was 0 seconds long but needs to be at least 3 - abandoning
== Spawn extension (incoming, 5008, 2) exited non-zero on ‘SIP/5009-09f4cbd8’
if you need any thing else please let me know.
Thanksin advance