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I have two MP-108 boxes working fine in
both ways. You have to check these; -
Assign
endpoint phone numbers to FXO ports (Note these numbers you will use in
extensions.conf later) -
Route
calls to asterisk in “Tel to IP Routing” table. -
Add endpoint
phone numbers to asterisk’s extensions.conf So whenever a call comes Audiocodes will
dial “endpointphonenumber (at) asterisk”. From: Mahilal Silva
[mailto:asteriskcrazy (at) gmail.com] Not quite sure.
Audiocodes gives a dialtone when the number is called from PSTN. After few
seconds I see the SIP invite to the Asterisk box. Asterisk responds with SIP
404 . On 6/12/06, Erick
Perez <eaperezh (at) gmail.com>
wrote: So is the problem with your audiocodes or with the asterisk system? |
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