[Asterisk-Users] Troubleshooting Asterisk

Paul Brock Paul at westcomuk.com
Tue Dec 21 04:44:15 MST 2004


>Great :-)

>If you use context=from-sip in sip.conf, you should include the [voiptalk]
>context into your [from-sip] context. (in the extension.conf)

>eg.

>[from-sip]

>include => 2001
>include => 2002
>include => voiptalk

>This way the Cisco's can call eachother, and dialout using the
dial->patterns defined in [voiptalk]

Done. Doesn't want to work (bearing in mind that I'm behind a NAT, I presume
I need to activate the NAT=yes settings. Also are there any ports that need
to be allowed through the NAT firewall at all? I could always just DMZ the
box....

Called number is 01934830055. The trace I'm seeing is as follows...

Using latest request as basis request
Sending to 192.168.1.151 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.151:17084
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1
(g723)
Found user '2001'
Looking for 01934830055 in from-sip
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6bd504eb
From: "2001" <sip:2001 at 192.168.1.150>;tag=000e833cb157000b51278729-524afec6
To: <sip:01934830055 at 192.168.1.150>;tag=as582ddfdc
Call-ID: 000e833c-b157000c-280d19df-5ea7d02f at 192.168.1.151
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:01934830055 at 192.168.1.150>
Content-Length: 0


 to 192.168.1.151:5060


Sip read:
ACK sip:01934830055 at 192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.151:5060;branch=z9hG4bK6bd504eb
From: "2001" <sip:2001 at 192.168.1.150>;tag=000e833cb157000b51278729-524afec6
To: <sip:01934830055 at 192.168.1.150>;tag=as582ddfdc
Call-ID: 000e833c-b157000c-280d19df-5ea7d02f at 192.168.1.151
Date: Tue, 21 Dec 2004 11:11:01 GMT
CSeq: 102 ACK
Content-Length: 0


8 headers, 0 lines
Destroying call '000e833c-b157000c-280d19df-5ea7d02f at 192.168.1.151'

Any thoughts? I've for the [voiptalk] Iax information in iax.conf..

[voiptalk]
type=peer
username=username
secret=password
host=217.14.132.162	;ip addr for voiptalk

Paul
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041221/b7289b66/attachment.htm


More information about the asterisk-users mailing list