[asterisk-users] What don't I get about SIP?
list at virtutel.ca
Fri Sep 8 13:50:24 MST 2006
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
Using INVITE request as basis request -
101e3648-dafdbf9a-e15173ad at 192.168.1.200
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
From: "CAP" <sip:000f42056d58-1 at test.test.ca>;tag=DAD6C20C-68263D4F
To: <sip:965 at test.test.ca;user=phone>;tag=as4db2b55c
Call-ID: 101e3648-dafdbf9a-e15173ad at 192.168.1.200
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:965 at 18.104.22.168>
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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?
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> 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
> 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?
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tim St.
> 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?
> 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?
> Tim St. Pierre
> IP telephony specialist
> Toronto: 647 722 6930
> Toll-Free 1 888 488 6940
> tim at communicatefreely.net
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> asterisk-users mailing list
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Silly idea, why don't you sniff the packets being sent over port 5060?
You'll be able to verify the conversation taking place.
rushowr at phreaker.net
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