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

[Mike]

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