Astersik Now realtime

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

can any one help?

I really need this resolved please come forward

Possibly, but they are all reading the AsteriskNow forum.

Can you please elaborate , what do you ,mean ?
Should I post this in AsteriskNow forum. instead of asterisk support


As it says in the instructions for the current forum, AsteriskNow questions should be asked on the AsteriskNow forum.

Alternatively, you need to reduce the test case to one that doesn’t depend on AsteriskNow.

The CLI trace is confusing, because it shows some successful calls, as well as the one that got busy.

One would need a trace of the SIP dialogue to understand why Asterisk thought 5008 was busy. From just the trace, it is possible that 5008 hasn’t put the phone down, yet, after calling 5009.

can you please advise me on how to generate CLI trace that can generate more detailed logs
One more question
How to handle NAT issue for Real time environment,
Do you have detailed instructions

Thanks again for coming forward.

sip set debug on

Never used real time. What have you tried and how did it go wrong?

Don’t expect to get detailed instructions on a peer support forum like this.

What I am trying to do is
I have all the realtime Tables setup in my sql database
I have created Record in SIP buddies and extensions tables
Now I want these Extensions to interact for inbound and out bound calls
but as soon as I place a call there is no ring tone on the outbound side on the ear piece and it goes to voice mail directly and there is no Ring on the in bound end .At this point there is no NAT involved this is all happening in my local network

I am using xlite as client in both computers


i only know that “SIP/5008-09f8e730 is circuit-busy” , msg displays
mostly when client is unreachable, or cannot be accessed.