SIP 2.0 not found?

Asterisk Asterisk Support
1

Hello

I use trunk called test_halonet_test (SIP). Outgoing connections works fine.
Incoming not. When i try to call that number, asterisk (sip debug) shows:
(217.11.128.5 and 217.11.128.50 are my provider ip addresses, MY_CELLPHONE is my cellphone number which i used to call, 48223492739 is my number which i am trying to call)

<-- SIP read from 217.11.128.5:5060:
INVITE sip:pagein@83.13.242.85:5060 SIP/2.0
Record-Route: sip:48223492739@217.11.128.5;ftag=5371A30-60D;lr=on
Via: SIP/2.0/UDP 217.11.128.5;branch=0
Via: SIP/2.0/UDP 217.11.128.50:5060;rport=49386
From: “Wywolanie” sip:MY_CELLPHONE@217.11.128.50;tag=5371A30-60D
To: sip:48223492739@217.11.128.5
Date: Wed, 25 Apr 2007 14:41:19 GMT
Call-ID: DD6B742D-F27111DB-AE6FD27E-6F91C5CB@217.11.128.50
Supported: timer
Min-SE: 1800
Cisco-Guid: 3714606797-4067496411-2926367358-1871824331
User-Agent: HaloNet GW
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID: “Wywolanie” sip:MY_CELLPHONE@217.11.128.50;party=calling;screen=yes;privacy=off
Timestamp: 1177512079
Contact: sip:MY_CELLPHONE@217.11.128.50:49386
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 485

v=0
o=CiscoSystemsSIP-GW-UserAgent 6937 1107 IN IP4 217.11.128.50
s=SIP Call
c=IN IP4 217.11.128.50
t=0 0
m=audio 16670 RTP/AVP 18 4 2 98 99 100 101 19
c=IN IP4 217.11.128.50
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:2 G726-32/8000
a=rtpmap:98 G726-24/8000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=direction:active

— (22 headers 20 lines) —
Using INVITE request as basis request - DD6B742D-F27111DB-AE6FD27E-6F91C5CB@217.11.128.50
Sending to 217.11.128.5 : 5060 (NAT)
Found peer 'trunk_halonet_out’
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 100
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port 217.11.128.50:16670
Found description format G729
Found description format G723
Found description format G726-32
Found description format G726-24
Found description format G726-16
Found description format X-NSE
Found description format telephone-event
Found description format CN
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x111 (g723|g726|g729)/video=0x0 (nothing), combined - 0x111 (g723|g726|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for pagein in DID_trunk_1 (domain 83.13.242.85)
Reliably Transmitting (NAT) to 217.11.128.5:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.11.128.5;branch=0;received=217.11.128.5
Via: SIP/2.0/UDP 217.11.128.50:5060;rport=49386
From: “Wywolanie” sip:MY_CELLPHONE@217.11.128.50;tag=5371A30-60D
To: sip:48223492739@217.11.128.5;tag=as18323428
Call-ID: DD6B742D-F27111DB-AE6FD27E-6F91C5CB@217.11.128.50
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

I use asterisk 1.2.14.
Is it the problem in “SIP/2.0 404 Not Found” ?
How to solve this problem ?

Thanx

2 
INVITE sip:pagein@83.13.242.85:5060 SIP/2.0 
...
To: <sip:48223492739@217.11.128.5>

This call will be routed to pagein, not to the number you’re expecting. This is what Asterisk is telling you: Looking for pagein in …
As soon as pagein is not present in your dialplan you’re receiving Not Found.

You should create the rule in your extensions.conf which will extract the number from To: field and then use that number for further routing.

3

In extensions.conf i have one context for all incoming connections:

[incoming]
exten => s,1,Dial(SIP/005)

which for all incoming connections redirect call to internal 005 number.
So - it should work, and work for all incoming numbers
(except the one i provided here)

Why ?

4

I want to add only that i do not have problems with incoming IAX connections, but with incoming SIP connectios. How should my sip.conf look like for incoming connections ?
Should i do any changes ? (if outgoing connection works fine) ?

I am almost sure that the problem is not with NAT/connection. (all packets arrive to my server)

Thanx