Outbound SIP

Get help with installing and running AsteriskNOW.

Moderators: Moderator, Support

Outbound SIP

Postby webjamm » Wed Jan 24, 2007 3:46 pm

Background:

My softswitch is broadsoft.
Installed asteriskNow on dell server
Configured service provider with IP of proxy to 192.168.1.1
Configured inbound route

Inbound calls from my broadsoft switch work great.

I set up outbound calling rules for 9-10 or more digits.


When I try to place a call from a phone configured behind the asterisk box, the call fails with 404 not found, but the asterisk box never sends the call to the IP that is configured in the service provider config.

This line in the debug specifically is confusing.

Looking for 4047900852 in default (domain 192.168.1.10)

Why does asterisk not send the call to the 192.168.1.1 IP when it doesnt find it locally?


Below is the debug output.



<--- SIP read from 192.168.1.24:52187 ---> INVITE sip:4047900852@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060
From: "James Webb" <sip:6000@192.168.1.10>;tag=000d285e9f2d00240431bed8-07afa245
To: <sip:4047900852@192.168.1.10>
Call-ID: 000d285e-9f2d003e-5690de11-190c28c3@192.168.1.24
Date: Wed, 24 Jan 2007 21:33:27 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:6000@192.168.1.24:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 247
Accept: application/sdp

v=0
o=Cisco-SIPUA 21111 24856 IN IP4 192.168.1.24 s=SIP Call c=IN IP4 192.168.1.24 t=0 0 m=audio 16460 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (13 headers 11 lines) ---
Sending to 192.168.1.24 : 5060 (no NAT)
Using INVITE request as basis request - 000d285e-9f2d003e-5690de11-190c28c3@192.168.1.24
Found no matching peer or user for '192.168.1.24:52187'
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 192.168.1.24:16460 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.1.24:16460 Looking for 4047900852 in default (domain 192.168.1.10)

<--- Reliably Transmitting (no NAT) to 192.168.1.24:5060 ---> SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.24:5060;received=192.168.1.24
From: "James Webb" <sip:6000@192.168.1.10>;tag=000d285e9f2d00240431bed8-07afa245
To: <sip:4047900852@192.168.1.10>;tag=as3e5aa42d
Call-ID: 000d285e-9f2d003e-5690de11-190c28c3@192.168.1.24
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000d285e-9f2d003e-5690de11-190c28c3@192.168.1.24' in 32000 ms (Method: INVITE) cs1*CLI>
<--- SIP read from 192.168.1.24:52188 ---> ACK sip:4047900852@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.24:5060
From: "James Webb" <sip:6000@192.168.1.10>;tag=000d285e9f2d00240431bed8-07afa245
To: <sip:4047900852@192.168.1.10>;tag=as3e5aa42d
Call-ID: 000d285e-9f2d003e-5690de11-190c28c3@192.168.1.24
Date: Wed, 24 Jan 2007 21:33:27 GMT
CSeq: 101 ACK
Content-Length: 0
webjamm
Newsterisk
 
Posts: 1
Joined: Wed Jan 10, 2007 1:51 pm

Return to AsteriskNOW Support

Who is online

Users browsing this forum: No registered users and 1 guest