[Asterisk-Dev] Sip Registration Bug?

[Ben Miller]

Agreed,
There are several ways that we can work around this, but the fact that
asterisk creates the inbound SIP channel as if it were the first
registration IS an incorrect behavior.

In addition, the fact that the second sip friend definition breaks both
friends tells me there is something in the code that is not allowing
full separation of the two channels registered with the same provider.

I will wrap up the details and submit a bug report on Mantis.
Thanks for the input.

Ben

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Preston
Garrison
Sent: Monday, March 07, 2005 3:08 AM
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] Sip Registration Bug?

The bug is simply this, instead of setting EXTEN to proper extension 
being dialed, it instead always is the first number plus some random 
garabage on the end :)


Preston Garrison
direct: 877-748-4142
fax: 310-774-3901
cell: 623-748-4140

-----Original Message-----
From: Greg Hill <gregh-asteriskd at hillnet.us>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Sent: Sun, 6 Mar 2005 22:42:18 -0700 (MST)
Subject: Re: [Asterisk-Dev] Sip Registration Bug?

Where's the bug? If you only tell asterisk which context to dump a call
into, then all registrations which result in calls being sent there will
go to the s extension and get handled by the same extension logic. The 
fix
is to give asterisk more information by specifying an extension to
override the s-extension default.

The fact that the OP points out that calls to his 800 got labeled as
SIP/888... instead does raise some suspicion.. does that also happen 
with
/exten appended to the register command? I agree that this behavior has
the appearance of a bug (to my untrained eye).

Greg


On Sun, 6 Mar 2005, Preston Garrison wrote:

> Actually that does work, still a bug though :)  But atleast that is a
> work around..
>
>
> Preston Garrison
> direct: 877-748-4142
> fax: 310-774-3901
> cell: 623-748-4140
>
> -----Original Message-----
> From: Greg Hill <gregh-asteriskd at hillnet.us>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Sent: Sun, 6 Mar 2005 21:33:09 -0700 (MST)
> Subject: Re: [Asterisk-Dev] Sip Registration Bug?
>
> > Summary:
> > When I register with two Broadvoice accounts, calls for the second
> > account registered come into asterisk as if for the first account.
> The
> > channel is actually named based on the first account.
> >
> > For example.  I have two accounts:  The numbers for this example 
are:
> > 8885551212 and 8005551212.
> >
> > In sip.conf I have:
> >
> > Externip=192.168.1.1   ;this is a real address in my real file
> > dtmfmode=inband
> > context=broadvoice-in  ; Default context for incoming calls
> >
> > register => 8885551212:password at sip.broadvoice.com
> > register => 8005551212:password at sip.broadvoice.com
> >
> >
> > When I dial the 8005551212 account I get an inbound call labeled:
> > SIP/8885551212-asdf.
> > The only way I can distinguish between them is to do a
> > SIPGetHeader(TO,To) and parse the variable.
>
> What if you add /exten to the end of your register statements? (if I
> understand it correctly) that should cause calls related to that
> registration to go to the specified context and extension, rather than
> going to the s extension. That would enable you to handle the 
different
> types of calls in the same context but with different command 
sequences.
> Or maybe you already thought about that and I misunderstand the 
problem
> you're experiencing..?
>
> Greg
>
>
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