[asterisk-users] What don't I get about SIP?

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

Mike

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
Via: SIP/2.0/UDP
192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45
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 65.111.23.42>
Content-Length: 0

-----Original Message-----
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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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: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] 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://5101@communicatefreely.net
> Toronto: 647 722 6930
> Toll-Free 1 888 488 6940
> tim at communicatefreely.net
> 
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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.

- --
S McGowan
VoIP Consultant
rushowr at phreaker.net

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