[Partysip-dev] Re: partysip config

[luke dobson]

>
> On Tue, 16 Jul 2002, luke dobson wrote:
>
> > Hi
> > I am using trying to use partysip with opensip to map phone numbers to
> > phones.  I have terminals such as address@hidden and
> > address@hidden  Dialling the entire IP, the phones can call each
> > other.  However I would like to dial 10 and get address@hidden  I
> > have played with the partysip.conf file and the user section looks like
> > this.
> >
> > user  sip:address@hidden   10    aa
> > user  sip:address@hidden   11   aa
>
> You can't do that. The url given for a user is made of his
> username @ domain.org where "domain.org is either 'serverip'
> or 'servername' from the partysip.conf.

Oh I see, this is basically for user authentication not any sort of routing.

> The real IP of user is received in the contact of REGISTER messages.

That makes sense.

> > authorisation is presently switched off
> >
> > When I call between the phones using the IP, I can see in the log that
> > the from: and to: fields are populated but when I dial the number alone
> > the to: field has a missing field.
> >
> > From:  <sip:address@hidden;user=phone>;tag=2908867716
> > To:  <sip:;user=phone>
>
> Is this message coming from your application? I don't see
> any reason why I should accept such a message.

Its coming from a Cisco ATA.

> > Further down the log contains the message
> >
> > | ERROR | <../../fsm/sipevent.c: 72> could not parse message
> > | ERROR | <udp.c: 220> udp module: Could not parse response!
> >
> > and partysip prints
> >
> > <msg_parser.c> error in msg_startline_init()
>
> the start line is the initial line of the request or response.
> You probably has the same value than in the "To" header above.
>
> > I assume I have done something silly in the config but what?  Any help
> > much appreciated!
>
> Don't think it's related.

I have it working now.  I think the ATAs were incorrectly configured for
SIP.  Thanks for your help with the config file anyway.

> > TIA
> > Luke Dobson

Cheers
Luke Dobson