jag

Nice feature! When you request a call back in 5 minutes, the phone rings and just hangs up, is this correct? or am I missing some wav files?

The 'Tones' drop down list on System -> General tab is empty. How does the PBX populate this tones list? does it read the audio_xx directories found in the main PBX directory?

This can be quite complex, so lets just look at connecting up 1 site for the moment. 1. Create a trunk between site A and B and between B and A 2. In the dial plan on Site A, create a pattern match to grab all calls starting with 50 and 5 digits long, if found route them to the Site B trunk. 3. On Site B, use the pattern matching in the 'extension' field of the trunk to send the call to the specific extension. For pattern matching look at the Wiki for a full explanation of how to do this http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk Repeat the process for all other sites. NOTE: Inbound calls on trunk is dependant on how you set-up your PBX. If you use multiple domains you should use the tel:alias feature, if they are in one domain, then you can simply use the vanilla extension number to route the call directly to the extension. You could also do this with DNS and not bother with a registered trunk. There are many different ways to do this, it all depends on how you want configure the PBX

How can I change the auto provisioning files on PBXNSIP? I need all the Snom and Polycom files. Thanks jag

How do you get round the problem where we use a global trunk AND all DIDs on one domain need to be the same? You can only have 1 tel:ALIAS on the system? This is standard stuff for some companies, however in an ITSP configuration we may need the facility of allowing all (or some) DIDS to be presented as 1 DID only.

We need to use the facility of $2 from the second user parameter. Are you saying we can't use this anymore. AND What about the $2 in the INVITE, is this a BUG?

The Outbound Trunk ANI Facility is not working. We add $2 in the Explicit Remote-Party-ID field. PBXNSIP does this INVITE sip:telephone_number@domain.com;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-e570ebb78b789357fbb229d06f25958d;rport From: "10002 10002" <sip:$2@trunk_domain.com>;tag=10450 This stops the CLI from being presented. ALSO You have changed the way trunks use the Remote Party/Privacy Indication: You have removed the straight RFC3325 option. In the example above the name "10002 10002" would be shown as the ANI and NOT the actual ANI which is stored in $2. Did it show "10002 10002" because of the $2 bug?