[asterisk-users] Panasonic x Asterisk if I can emulate Panasonic fast!

[C F]

On Thu, Oct 16, 2008 at 8:45 AM, Olivier <oza-4h07 at myamail.com> wrote:
>
>
> 2008/10/16 C F <shmaltz at gmail.com>
>>
>> * Live call screening - Yes there is a hack that can do it, but it's a
>> hell of a hack.
>> * Phones that can do most of the usefull features supported by the PBX
>> for a reasonable price with LED buttons, including the following
>> features:
>> ** Call recording with LED indication, while at it, the recordings
>> integrate seamlessly with your voicemail, which means you don't need
>> to browse the file system on the PBX to listen to it.
>
>
> What would be missing to integrate this feature ?
> With features.conf, it should be possible to map key combinations to an
> Asterisk application (maybe an AGI script ?)
> From there, it should be possible to drive SIP hardphones BLF status, don't
> you think ?

Yes and no, the real thing would be to be able to get a status
feedback from Asterisk that it's actually recording, then based on
that one would be able to use devstate from 1.6 to turn on BLF.
For where to store the recordings, the best way would be if arg could
be passed to the recording app indicating to which voicemail user to
send it and follow the settings (email, pager etc.) for that user.

>
>>
>> ** Login/Logout of queues, Day/Night mode buttons with indication (1.6
>> has this as well).
>> ** Company internal directory on the phone updated on the PBX
>
>  Some (most ?) IP phones support this

The phones support it, but not from asterisk, what that requires is a
separate provisioning system for each type of phone, that pulls the
data from central database, not that hard to build and maintain in
theory, just very costly to develop since such a provisioning system
doesn't yet exist, at least AFAIK.

>>
>> ** System Speed Dial on the display updated by the PBX
>
> This one is interesting.
> I can't see a way to do it.
> Ant idea ?

As far as SIP goes, no it's impposible, however as far as end
users/admins are concerned the PBX is/could also be the system
provisoining the phones, in which case what I wrote above could be
done.

>>
>> ** Call Fwd by PBX with LED indication (not phone based callfwd which
>> sucks).
>
> Some IP phones support this

Which ones?

>
>>
>> ** On screen Voicemail (on the phone).
>
> high end ip phones (XML) should support

Again XML, expensive development costs.

>>
>> ** Line assignment to buttons with LED indication, and hold indication.
>
> For this one, I don't know. SCA, maybe ?
>>
>> ** Hold ringback (some IP phones support it).
>> There are many more features but I can't remember them at the moment.
>>
>> Granted in bigger installations there many more factors and usually
>> more funding which makes the above list almost obsolete for the
>> features that Asterisk does have.
>>
>> Again my advice do not go with Asterisk for this installation go with
>> Panasonic.
>>
>>
>>
>>
>> >> What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys
>> >> SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured
>> >> Asterisk/Fedora 9 so I can make SIP->PSTN and PSTN->SIP calls.
>> >>
>> >> Works. Now, I need this help, please:
>> >>
>> >> * Dialing from inside (pap2-FXS connected phone) to another number on
>> >> the same city (goes out by SPA3102 FXO), voice works fine. But when a
>> >> menu answers, and I dial over, the menu dialed keys works only 20% of
>> >> all times. Why could this would be? Voltage levels? sound gains? Dialed
>> >> keys get distorsioned when passing over the 2 Linksys? Linksys or
>> >> Asterisk swallowing some dialed key? I noticed some echo...
>> >>
>> > Probably you are sending dtmf signals inband. Try outband.
>> > For the echo, try to change the FXO/FXS impedance, and/or playing with
>> > the rx and tx gains. I assume that do you have echo cancelling enable in
>> > both SPA.
>> >> * I need to assign two codes to each user, one for international calls
>> >> charged to the office, another for international calls charged to the
>> >> user. If the user enters an incorrect code, the call should not
>> >> proceed.
>> >>
>> > See account codes. You can start here:
>> > http://www.voip-info.org/wiki-Asterisk+Billing
>> >
>> >> * I need to get a formatted calls report for the administrators to
>> >> charge the users.
>> >>
>> > See same link, or google for billing
>> >> I just am confused and stucked with all the documentation in Internet,
>> >> and all this new asterisk jargon. I just need some links (or some
>> >> directions) to go fast on this topics. Of course, some more help would
>> >> be appreciated.
>> >>
>> > The link to start:
>> > http://www.voip-info.org
>> >
>> >> Thanks a lot.
>> >>
>> > De nada
>> >
>> > Jorge
>> >
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