Hi Noah,
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Noah Miller > Sent: Thursday, December 13, 2007 21:02 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk 1.2.18 and Polycom > phones notforwarding anymore > > Hi Mick - > > > I've had a functioning Asterisk system (1.2.18), which I haven't > > reconfigured in any way, that is just now refusing to > forward calls. I > > only have Polycom phones. When I use the phone's forward feature > > (forwarding the phone with extension 204 to extension 206, > which used > > to work as recently as yesterday) I get this in the > console: "called > > sipreg-12344". No ringing, nothing. Just a long silence while the > > Dial cmd times out. > > > > I`ve rebooted the phones, the router, everything in fact, > but no result. > > Would anyone have an idea where to look next? > > I'd enable verbose logging and see what you can find there. To do so: > > 1. Edit logger.conf > 2. add the word "verbose" to the line "messages =>" (and make > sure the line is uncommented) 3. restart asterisk > > Check it out to see what's going on. I don't get much more than the CLI shows. SIP reg reg_a is the line called, reg_b is the line that a is redirected to on the phone (using the line forward feature of my Polycoms 650 or 501). I do get a "Called Reg_a" message in the log, but that's it. No reference to reg_b. I guess SIP debugging would help more. This is what I get between the Dial cmd and the timeout (25 seconds, as you can see from the dial command). It's like reg_a gets the call, and hold on to it for no reason. As I said, it works fine when the forward is removed (i.e. reg_a rings), and it worked fine before today. -- Executing Dial("SIP/5060-0970fbd8", "SIP/reg_a|25") in new stack We're at 56.45.32.12 port 17404 Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 11 lines Reliably Transmitting (NAT) to 44.67.87.98:5060: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 56.45.32.12:5060;branch=z9hG4bK493a82e2;rport From: "Joe Smith" <sip:[EMAIL PROTECTED]>;tag=as36ef9642 To: <sip:[EMAIL PROTECTED]:5060> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 14 Dec 2007 02:12:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 239 v=0 o=root 9207 9207 IN IP4 56.45.32.12 s=session c=IN IP4 56.45.32.12 t=0 0 m=audio 17404 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called reg_a hd-t3143cl*CLI> <-- SIP read from 44.67.87.98:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 56.45.32.12:5060;branch=z9hG4bK493a82e2;rport From: "Joe SMith" <sip:[EMAIL PROTECTED]>;tag=as36ef9642 To: <sip:[EMAIL PROTECTED]:5060>;tag=3135D762-658B9D03 CSeq: 102 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]:5060> User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078 Content-Length: 0 -- Nobody picked up in 25000 ms Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms Reliably Transmitting (NAT) to 44.67.87.98:5060: CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 56.45.32.12:5060;branch=z9hG4bK493a82e2;rport From: "Joe Smith" <sip:[EMAIL PROTECTED]>;tag=as36ef9642 To: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 102 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Thanks so much, Mick _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
