[asterisk-users] Stuck somewhere - INVITEs ignored?!

[Sascha Pollok]

Dear folks,

I have set up a new Asterisk server running 1.4.0 and a SNOM 360
sip-client (also tried Eyebeam). I have configured some dozens SIP
clients on 1.2 so I am wondering why the phone is not able to place
an outgoing call. Here is the relevant (guess so) sip.conf part:

[2899]
type=friend
secret=2899
context=pbx
host=dynamic
nat=no
allow=all

The phone registers properly, the context pbx contains a simple
extension (answer, musiconhold) that I am trying to call. Now when
the phone tries to dial this extension, this is what happens:

<--- SIP read from MY_PHONES_IP:2051 --->
INVITE sip:800 at ASTERISK_SERVERS_IP SIP/2.0
Via: SIP/2.0/UDP MY_PHONES_IP:2051;branch=z9hG4bK-l009xucwo4bl;rport
From: "Name" <sip:2899 at ASTERISK_SERVERS_IP>;tag=z3lofcfvnd
To: <sip:800 at ASTERISK_SERVERS_IP>
Call-ID: 3c2762b76b6c-er1hxx3ca3fu at snom360
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:2899 at MY_PHONES_IP:2051;line=pysdpam9>
P-Key-Flags: resolution="31x13", keys="4"
User-Agent: snom360/3.60i
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces
Session-Expires: 3600
Proxy-Authorization: Digest
username="2899",realm="asterisk",nonce="49175a6d",uri="sip:800 at ASTERISK_SERVERS_IP",response="xxx",algorithm=md5
Content-Type: application/sdp
Content-Length: 372

v=0
o=root 758418159 758418159 IN IP4 MY_PHONES_IP
s=call
c=IN IP4 MY_PHONES_IP
t=0 0
m=audio 56202 RTP/AVP 0 8 9 2 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

<------------->
--- (18 headers 17 lines) ---
Ignoring this INVITE request

[Jan 11 11:15:50] NOTICE[5144]: chan_sip.c:13534 handle_request_invite:
Unable to create/find SIP channel for this INVITE

<--- Transmitting (no NAT) to MY_PHONES_IP:2051 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP
MY_PHONES_IP:2051;branch=z9hG4bK-l009xucwo4bl;received=MY_PHONES_IP;rport=2051
From: "Name" <sip:2899 at ASTERISK_SERVERS_IP>;tag=z3lofcfvnd
To: <sip:800 at ASTERISK_SERVERS_IP>;tag=as4af51482
Call-ID: 3c2762b76b6c-er1hxx3ca3fu at snom360
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:800 at ASTERISK_SERVERS_IP>
Content-Length: 0



So basically, the INVITE request is ignored. I even searched through
chan_sip.c trying to find out why SIP_PKT_IGNORE is set but got lost
somewhere. I guess it is some easy thing with domains, IPs, whatever
but can someone please point me into the right direction?

Thank you very much.

Cheers
Sascha