-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Mike wrote: > It's not a silly idea, I've been doing some sniffing and debugging with my > limited knowledge of the whole process. I found this in the debug stream > after having dialed "965"). > > Notice this line: SIP/2.0 484 Address Incomplete. > > Is this what I was suspecting, that it knows it could match a pattern > (_9XXXXX) with a few more digits and so waiting for those digits from the > user? How can I disable this or turn it off? The Polycom 501 "supports 484 > responses", but how can I either: > 1) Disable it in the phone > 2) Disable it in Asterisk > > Mike > > > > > > > > > > Using INVITE request as basis request - > [EMAIL PROTECTED] > Sending to 192.168.1.200 : 5060 (NAT) > Found user '000f42056d58-1' > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 18 > Found RTP audio format 101 > Peer audio RTP is at port 192.168.1.200:2228 > Found description format PCMU > Found description format PCMA > Found description format G729 > Found description format telephone-event > Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c > (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) > Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Looking for 965 in context_a (domain test.test.ca) > Reliably Transmitting (NAT) to 45.67.312.45:5060: > SIP/2.0 484 Address Incomplete > Via: SIP/2.0/UDP > 192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45 > From: "CAP" <sip:[EMAIL PROTECTED]>;tag=DAD6C20C-68263D4F > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as4db2b55c > Call-ID: [EMAIL PROTECTED] > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr > Sent: September 8, 2006 4:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] What don't I get about SIP? > > Mike wrote: >> Thanks Tim. > >> I've been trying to find out what's happening. Basically, somehow, it >> seems that my Polycom 501 knows what extensions are valid and which >> aren't in my dialplan. Obviously, the 501 doesn't really know that, >> but Asterisk seems to return it this info (sort of :"valid", "invalid" >> or "could be valid, need more digits to know") when I press send. > >> I know it sounds mad, and I would love nothing more than being told I >> am an idiot because or x and y. Why do I feel that this info is >> passed from Asterisk to the 501? > >> Well, take the following (very simple) dialplan > >> [context_a] >> Exten => 1234,1,Noop(foo) > >> Exten => _9XXXX,1,Noop(bar) > >> Exten => i,1,Noop(invalid) > > >> What happens when I dial out is the following: > >> 1) 1234: Noop(foo) ; good > >> 2) 444444444: A congestion tone is heard from the phone (but >> Asterisk's CLI doesn't show anything...no "sent into invalid extension >> '444444444' in context 'context_a', but no invalid handler > >> 3) 934 : It's invalid, but it could match the pattern is I added some >> digits. I expect an invalid extension message, but what actually >> happens is the phone tries the send something (I can see an icon >> moving on the phone) but the phone stays quiet (no stuttering tone or >> whatever). It waits, I can input more digits on the phone. > >> Let's just take 1) and 2). Why is Asterisk not going into the i >> extension like it should? > >> Mike > > > > > > >> -----Original Message----- >> From: [EMAIL PROTECTED] >> [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. >> Pierre >> Sent: September 8, 2006 2:54 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [asterisk-users] What don't I get about SIP? > >> With SIP, asterisk processes the digits it receives in the invite from >> the Polycom. > >> There is no communication of dialplan information in SIP. The polycom >> should send the digits as soon as you press dial. You can program the >> polycom with a dialplan that will tell it when to send the digits, but >> that only works if you dial off-hook. I like on hook dialling, since >> it sends what i tell it, when I tell it. This should never happen >> when you press dial - it should try right away. My 301 does this, >> maybe they changed something in the newer firmware? > >> -Tim > >> On September 8, 2006 14:33, Mike wrote: >>> I've been running into an issue with my Polycom 501 and Asterisk. >>> >>> I realized, after much mucking around, that when I dial a number (and >>> press the send key) that is invalid , but could still match an >>> Asterisk pattern >>> (example: I dial 567, which is not a valid extension, but my diaplan >>> accepts _567XXXX as a pattern) instead of sending the call as is and >>> ultimately failing, the phone is "intelligent enough" to sit and wait >>> for extra digits in case I meant to dial 567111. >>> >>> Now thats a problem for me. How can I make Asterisk (or the 501) >>> treat the attempted extension 567 as a valid try and let Asterisk >>> handle the error ?(instead of the phone trying to do what it think is >>> best and handling the error on it's own). >>> >>> Is there an Asterisk setting for that? >>> Failing that, is there a Polycom setting to disable this "intelligent" >>> error handling? >>> >>> >>> Mike >> -- >> Tim St. Pierre > >> IP telephony specialist >> sip://[EMAIL PROTECTED] >> Toronto: 647 722 6930 >> Toll-Free 1 888 488 6940 >> [EMAIL PROTECTED] > >> _______________________________________________ >> --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 > > Silly idea, why don't you sniff the packets being sent over port 5060? > You'll be able to verify the conversation taking place. > Sounds more like the phone's not sending a proper INVITE to me. If you like, you can send me a short trace of the traffic. I personally use either tcpdump or tethereal with a capture filter of "port 5060" and use the -w option to write the packets to a file so I can view it in more depth.
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