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:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP MY_PHONES_IP:2051;branch=z9hG4bK-l009xucwo4bl;rport > From: "Name" <sip:[EMAIL PROTECTED]>;tag=z3lofcfvnd > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > Max-Forwards: 70 > Contact: <sip:[EMAIL PROTECTED]: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:[EMAIL > PROTECTED]",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:[EMAIL PROTECTED]>;tag=z3lofcfvnd > To: <sip:[EMAIL PROTECTED]>;tag=as4af51482 > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[EMAIL PROTECTED]> > 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 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
