[asterisk-dev] New feature

[Piotr Goczal]

Dnia 24-10-2008, pią o godzinie 12:05 +0400, Sergey Tamkovich pisze:
Hi,

> What we should send back to phone's request? What should be the source 
> of data for such directory? any proposals?
It could be some external source and/or internal database used in
voicemail directory module.

For me (a the beginning) it could be plain text file.

Best regards

Piotr


> 
> Piotr Goczal wrote:
> > Dnia 24-10-2008, pią o godzinie 01:08 +0300, michel freiha pisze:
> > Dear Michel,
> >
> >   
> >> Can you please explain how can we benefit from this new feature?
> >>     
> > Sorry but I don't quite get the question. Is answer: "Linksys phones
> > will be better supported by Asterisk" a good answer?
> >
> > I'm using asterisk and Linksys (phones and some PSTN gateways) based
> > system and I like it. There are some drawbacks of such solution, one of
> > them is lack of common directory.
> >
> > I thought the it could be a good idea to implement such feature in the
> > asterisk. I agree that directory should be done in "normal way" by LDAP
> > but............. We can blame Linksys and don't use directory or just
> > write "ugly patch" and fix "their mistake".
> >
> > Best regards
> >
> > Piotr
> >
> >   
> >> Regards
> >>
> >> On Thu, Oct 23, 2008 at 10:39 PM, Piotr Goczal <bilbo at man.lodz.pl>
> >> wrote:
> >>         Hi,
> >>         
> >>         My name is Piotr Goczal and unfortuantelly I'm not a C/C++
> >>         coder.
> >>         Linksys SPA-9XX phones have nice feature called "Corporate
> >>         directory"
> >>         unfortunatelly it works only with SPA-9000 PBX. It looks that
> >>         it's done
> >>         via simple(?) Request INFO:
> >>         
> >>         ---
> >>         INFO sip:dir at 172.16.0.49:6060 SIP/2.0
> >>         Via: SIP/2.0/UDP 172.16.0.46:5060;branch=z9hG4bK-76687070
> >>         From: "Test2" <sip:23 at 172.16.0.49>;tag=1aaa997cfdd10d2do0
> >>         To: <sip:dir at 172.16.0.49>
> >>         Call-ID: 376cb484-ee5a72db at 172.16.0.46
> >>         CSeq: 2832 INFO
> >>         Max-Forwards: 70
> >>         User-Agent: Linksys/SPA922-6.1.3(a)
> >>         Content-Length: 0
> >>         ---
> >>         
> >>         and answer:
> >>         
> >>         ---
> >>         SIP/2.0 200 OK
> >>         To: <sip:dir at 172.16.0.49>;tag=60b7acd9-0
> >>         From: "Test2" <sip:23 at 172.16.0.49>;tag=1aaa997cfdd10d2do0
> >>         Call-ID: 376cb484-ee5a72db at 172.16.0.46
> >>         CSeq: 2832 INFO
> >>         Via: SIP/2.0/UDP 172.16.0.46:5060;branch=z9hG4bK-76687070
> >>         Server: Linksys/SPA9000-6.1.5
> >>         Allow-Events: talk, hold, conference, x-spa-cti
> >>         Content-Length: 136
> >>         Content-Type: application/directory
> >>         
> >>         FXS1;21
> >>         FXS2;22
> >>         Test2;23
> >>         Test1;24
> >>         Auto Attendant;aa
> >>         Internal Music;imusic
> >>         Page Group;pagegroup
> >>         Station;Test2
> >>         Station;Test1
> >>         wszyscy;8000
> >>         ---
> >>         
> >>         e.x. : "FXS1" - name of extension; 21 - extension number
> >>         
> >>         If anyone is interested in implementing it I can offer my help
> >>         with
> >>         testing. I've mini lab with SPA-9000, SPA-400 and some SPA-9XX
> >>         phones to
> >>         do some reverse enginering test and of course normal working
> >>         asterisk
> >>         1.4 + Linksys phones instalation.
> >>         
> >>         Best regards
> >>         
> >>         Piotr
> >>         
> >>         
> >>         
> >>         
> >>         
> >>         
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> >
> >
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