We are able to make external calls as well as make and receive internal calls.
We are running FreePBX 184.108.40.206 on Asterisk 1.8.5 with Polycom 331’s using SIP 3.3.1. Our gateways are Cisco 2951’s with dial peers defined as follows for our test extension, 8275:
dial-peer voice 4 pots
description Inbound calls from Integra
incoming called-number …
dial-peer voice 8275 voip
description Gateway to Asterisk
session protocol sipv2
session target sip-server
voice-class codec 1
Call Logs show:
21. 2012-01-24 07:20:17 SIP/148.18… 7753366400 7753366400 s ANSWERED 00:12
22. 2012-01-24 07:16:58 SIP/148.18… 7753317747 7753317747 s ANSWERED 00:07
Trunks in FreePBX are setup as:
Name=CiscoGateway; Outbound CallerID=“Washoe” <7753366400>; CID Options “Allow Any CID”; PEER Details are
Outbound Route ~
Trunk Sequence is CiscoGateway
Inbound Route for 8275 is:
DID Number: 8275
Destination: Extensions - 8275
Extension 8275 uses Inbound DID 8275
We do not use secrets at this time (ie blank)
deny & permit both equal = 0.0.0.0/0.0.0.0
Optional Destinations go to Voicemail for 8275
Digit Maps for the Polycom 331 templates are: [2-9]11|0T|9011xxx|*[2-9]xx|*80[2-9]xx|9[2-9]xxxxxxxxx|[2-9]xxxT; Timeouts = 3|3|3|3|3|3|3
We are not in a position to rollout all our phone lines at the same time. Only one extension at a time.
I have not been successful in receiving any results when searching subject content.
At this point, there isn’t any agreement as to whether this is an issue with my FreePBX settings, the Cisco router guys configs, or the TDM guys configs (using an archaic ASPA program designating and extension as a station or an LCR). Everyone is pointing to the other. ;[
If anyone can confirm or deny my setting are correct or incorrect or lacking something, I would be very very grateful. I no doubt have missed providing some pertinent info. So please feel free to request whatever you might need (ie an Asterisk log dump, which I can’t really make sense of).
Thank you for your help!