Hi Again Mick - OK stupid question time: Can you successfully make a call from ext 204 to 206? Are those IP's on the phones real? Has any of the IP routing changed? What does your sip.conf look like?
- Noah On Dec 13, 2007 9:21 PM, Mike <[EMAIL PROTECTED]> wrote: > 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 > _______________________________________________ --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
