Zamknięty

AsteriskNOW / FreePBX SIP Problem

We have a new AsteriskNOW 1.5 setup (Asterisk, FreePBX, etc).

Almost everything is working smoothly, but we have one problem with outgoing calls via SIP trunks. These calls are not hanging up correctly. EG:

** We call someone

** Their phone rings

** We hang up before they answer

** Our handset and the FreePBX admin panel show that the call has ended

** BUT their phone keeps ringing (until they answer, drop out, etc)

Incoming calls and calls made via our PSTN trunks are all fine. This is just a problem with outbound SIP - presumably a problem with the way our Asterisk box is communicating with our VoIP provider.

We have tried several 'tweaks' in the FreePBX Trunks screen, including changing to codecs recommended by our provider, etc, but nothing seems to work. Seems like it's probably time for someone to check through our config files.

WITH YOUR BID (in PM) - please indicate your level of experience with Asterisk/AsteriskNOW/FreePBX, an estimate for the number of hours this will take to complete, and an indication of your experience in resolving similar problems for other clients.

Thank you.

JD

Umiejętności: Linux, Administrator systemu

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O pracodawcy:
( 17 ocen ) Donvale, Australia

Numer ID Projektu: #415212