[asterisk-users] Asterisk 1.2.18 and Polycom phones notforwarding anymore

[Noah Miller]

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 <list at virtutel.ca> wrote:
> Hi Noah,
>
>
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] 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:reg_a at 44.67.87.98:5060 SIP/2.0
> Via: SIP/2.0/UDP 56.45.32.12:5060;branch=z9hG4bK493a82e2;rport
> From: "Joe Smith" <sip:5555551234 at 56.45.32.12>;tag=as36ef9642
> To: <sip:reg_a at 44.67.87.98:5060>
> Contact: <sip:5555551234 at 56.45.32.12>
> Call-ID: 45d35cd201d3e11c3872a99c4840cf4a at 56.45.32.12
> 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:5555551234 at 56.45.32.12>;tag=as36ef9642
> To: <sip:reg_a at 44.67.87.98:5060>;tag=3135D762-658B9D03
> CSeq: 102 INVITE
> Call-ID: 45d35cd201d3e11c3872a99c4840cf4a at 56.45.32.12
> Contact: <sip:reg_a at 44.67.87.98:5060>
> User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.1.2.0078
> Content-Length: 0
>
>
>     -- Nobody picked up in 25000 ms
> Scheduling destruction of call
> '45d35cd201d3e11c3872a99c4840cf4a at 56.45.32.12' in 32000 ms
> Reliably Transmitting (NAT) to 44.67.87.98:5060:
> CANCEL sip:reg_a at 44.67.87.98:5060 SIP/2.0
> Via: SIP/2.0/UDP 56.45.32.12:5060;branch=z9hG4bK493a82e2;rport
> From: "Joe Smith" <sip:5555551234 at 56.45.32.12>;tag=as36ef9642
> To: <sip:reg_a at 44.67.87.98:5060>
> Call-ID: 45d35cd201d3e11c3872a99c4840cf4a at 56.45.32.12
> CSeq: 102 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
>
> ---
>
>
> Thanks so much,
>
>
> Mick
>
>
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