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
