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

[Sascha Pollok]

Me again,

as a quick followup. Downgrading to SVN-branch-1.2-r50495 without
changing anything to the config-files immediately made all SIP
calls from the phone -> asterisk work. Did I miss something
in the changelog? :-/

Thanks for an advice
Sascha

On Thu, 11 Jan 2007, Sascha Pollok wrote:

> 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
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