Sorry, meant the 9116
Andre Courchesne
Concepteur Logiciel - Software Developer [EMAIL PROTECTED]
PrivalODC Inc.
9955 ave Catania, local 145
Brossard, QC
J4Z 3V5
Web.: http://www.prival.ca
Tel.: (450) 761-9973 poste 635
1-866-761-9973
Fax.: (450) 761-9842
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John Lange wrote:
The Aastra 9112i is a SIP phone so you must mean another model? In any
case, I would think the Aastra SIP "terminals" (as they call them) would
be just as durable as the analog only models.
If you are doing this in a student residence with analog phones, why not
just require them to supply their own phones?
The Call-waiting-caller-ID features is quite nice so I'd encourage you
not to disable it. Try testing it with a consumer level phone that
supports caller id to confirm it's working.
I have not encountered a current model of consumer phone with caller id
that does not also support callerID on call waiting.
--
John Lange
www.johnlange.ca
On Wed, 2008-10-22 at 16:38 -0400, Andre Courchesne - Consultant wrote:
I'm going through the audiocode config for that right now.
IP phones are not an option. This site has 14 MP124 devices and ip phones would
not survive 1 week (student residences).
Jim Van Meggelen wrote:
Sounds like it's sending CLID via FSK.
Either turn off CLID as part of call waiting (in the Audiocodes), or
replace the analog phone with an IP phone (which does not need to do
signalling in band). Also, the analog phone needs to be able to handle
CLID with call waiting, but even so it'll have to mute the audio path to
send the info, so that won't solve the problem of the audio cutting out..
Jim
Andre Courchesne - Consultant wrote:
Hi again,
In the following architecture:
Asterisk server ----- Audiocode MP124 ----- Analog phone
When an analog phone is on the line and he gets a second call, there
is a loud blip on the line and a subsequent mush less loud one. My
customer is complaining about the first one because it cuts
conversation. The sound is more like a blip-chirp sound.
I tried playing with the audiocode configuration (which is very
limited) and with the asterisk indications.conf to no result. Any ideas?
Thanks,
Andre
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