[asterisk-users] Help with sip configuration

[Joseph Jacobson]

On 07/08/08 11:55, Matt Riddell wrote:
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>Hash: SHA1
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>Ok, if you type:
>
>core set verbose 10
>
>and
>
>core set debug 10
>
>Then drop the file into /var/spool/asterisk/outgoing
>
>a) does the file disappear
>b) does anything come up in the console
>c) what is the date on the file i.e.:
>
>send us
>
>ls -alh /var/spool/asterisk/outgoing
>
>and
>
>date
>


Tried this and saw nothing in the console.  Thought this was telling
so I overwrote my modules.conf with the sample modules.conf.  Restarted.
Asterisk attempted to process the call file after this. 

Doesn't look like it picked up and processed the extension correctly.
The "To:" line in the sip dialog below doesn't include the number I'm
trying to dial.  Does that point to a problem(s) in my extensions.conf,
sip.conf, call file, or something else?


Thanks,

Joe

------------


     -- Attempting call on SIP/user1 for 2225552222 at outboundmsg1:1 (Retry 1)
Audio is at 10.52.161.72 port 21072
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.52.2.43:5060:
INVITE sip:sip.example.com SIP/2.0
Via: SIP/2.0/UDP 10.52.161.72:5060;branch=z9hG4bK46f3cbb5;rport
From: "asterisk" <sip:user1 at example.com>;tag=as396a7c42
To: <sip:sip.example.com>
Contact: <sip:user1 at 10.52.161.72>
Call-ID: 047b9d0d06e47fd247d95ae03076abd4 at example.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 08 Jul 2008 01:20:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 12778 12778 IN IP4 10.52.161.72
s=session
c=IN IP4 10.52.161.72
t=0 0
m=audio 21072 RTP/AVP 3 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from 10.52.2.43:38345 --->
SIP/2.0 100 Trying
From: "asterisk" <sip:user1 at example.com>;tag=as396a7c42
To: <sip:sip.example.com>
Call-Id: 047b9d0d06e47fd247d95ae03076abd4 at example.com
Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.52.161.72:5060;branch=z9hG4bK46f3cbb5;rport=5060
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 10.52.2.43:38345 --->
SIP/2.0 404 Not Found
From: "asterisk" <sip:user1 at example.com>;tag=as396a7c42
To: <sip:sip.example.com>
Call-Id: 047b9d0d06e47fd247d95ae03076abd4 at example.com
Cseq: 102 INVITE
Via: SIP/2.0/UDP 10.52.161.72:5060;branch=z9hG4bK46f3cbb5;rport=5060
User-Agent: sipX/3.6.6 sipX/registry (Linux)
Date: Tue, 08 Jul 2008 01:20:31 GMT
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, REGISTER, SUBSCRIBE
Accept-Language: en
Supported: gruu
Contact: sip:10.52.2.43:5070
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Transmitting (no NAT) to 10.52.2.43:5060:
ACK sip:sip.example.com SIP/2.0
Via: SIP/2.0/UDP 10.52.161.72:5060;branch=z9hG4bK46f3cbb5;rport
From: "asterisk" <sip:user1 at example.com>;tag=as396a7c42
To: <sip:sip.example.com>
Contact: <sip:user1 at 10.52.161.72>
Call-ID: 047b9d0d06e47fd247d95ae03076abd4 at example.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
       > Channel SIP/user1-09688c60 was never answered.