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

Tim St. Pierre tim at communicatefreely.net
Fri Sep 8 12:44:45 MST 2006

Now that is really odd.

Try sip debug peer (peername of the polycom)

This will let you see the sip packets go by when you do this, so you can see 
the responses it is, or isn't getting.

I'll have to look up the SIP response codes, but I do know that there is one 
for "not found" which should correspond with an invalid extension.  Because 
the call is not actually set up yet, asterisk will return a "not found" 
message rather than answer the call, only to direct it to an i extension.  
This is only used for calls already in progress.

I don't know if there is a sip response for "need more digits" or something 
like that.  Turning on the sip debug will tell you EXACTLY what the polycom 
is saying to asterisk, and vice versa.  Note: I like to hit scroll lock after 
I hit call, before I hangup so that it doesn't fill my screen up with all the 
cancel messages - that will put you just below the important parts of the 

On September 8, 2006 15:21, 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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Tim St. Pierre

IP telephony specialist
Toronto: 647 722 6930
Toll-Free 1 888 488 6940
tim at communicatefreely.net
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