deleting outgoing digits and incoming call route help

[rmattich]

I’m configuring a new install of 3cx V8. I have run into a couple of issues and wondered if there was a way to accomplish the following. I have my 3cx connected to an Avaya S8730 via sip trunk. I’m using 3 digit dial plan on the 3cx and 4 on the Avaya.

Users are used to dialing DID numbers into the Avaya that now can be reached via the sip trunk. I want to create an outbound rule on the 3cx that if a user dials a number xxx-xxx-xxxx the digits are dropped and replaced with the corresponding 4 digit extension. However, in the outbound rules, I only see the ability to delete a maximum of 9 digits. Why is 9 the max? Is there a way to change this?

Second question is on inbound routing. I created a VOIP connection on the 3cx for incoming calls from the Avaya and added a DID for that trunk. On the voip provider port page, I have to select where the call is routed. Of the limited options, the one that I selected is connect to extension. I have to select a static extension that all calls that come into that trunk are routed to. How can I get a call that comes in that trunk to route to the extension that was called? I see the correct extension in the INVITE, however the system ignores that and routes the call to the static extension specified in the voip did config.

 

[leejor]

How can I get a call that comes in that trunk to route to the extension that was called?

Did you look at the Inbound Rules options, there is a set-up for defining DID's

 

[rmattich]

yes, but it could be any one of 700 extensions coming in from the Avaya side calling any of 20 extensions on the 3cx side. I only can define one destination extension on each inbound route. How can Avaya extension 3423 call 3cx extension 101 or 102 or whatever extension is dialed. The sip invite shows the extension that the Avaya user dialed but is ignored and routed to the static extension set in the inbound route.

 

[leejor]

Did you look at this...
http://wiki.3cx.com/documentation/voip-providers/configuring-did-with-a-voip-provider

http://wiki.3cx.com/documentation/voip-providers/how-to-solve-source-identification-issues

The WiKi has a lot of info and your Avaya is just another VoIP provider to 3CX.

 

[rmattich]

I saw that documentation, however, it doesn't match my screen when I go to set it up. I don't have the part where I can add SIP field, unless I have to enable it somewhere else. Won't that just allow me to have multiple DID's associated with that VOIP provider route to one destination? The incoming call is being identified as far as I can tell, it just is routing to the one extension that I set in the inbound parameters. I'm just not seeing how I can route call to destination based on destination specified in the to field.

 

[leejor]

Are you using a paid or free version? I don't know if this is the reason but in the free version this (Establish Standards-based SIP Trunks with other SIP Servers) isn't included.

 

[rmattich]

I have the free version. Once I had it installed, there were several options that weren't enabled. I was then sent a key code to enter that I thought unlocked the same features as the paid version with the exception that it was limited to 2 at once. I have the sip confiugred and am able to receive/send calls, it is just that you have a PBX which would imply that you have more than one extension that you route calls to/from. I am not able to figure out how a remote end can call different extensions. The destination is statically set. Might as well just put a sip phone at the other end and bypass 3cx. I also am not clear on support. It seems that if you have to have the paid version for support. Can you not purchase a support package for the free version. If I can't get the free version working as I want, why would I want to purchase a license?

 

[leejor]

I'm afraid that someone else is going to have to "run" with this as I don't use, so consequently have not set up, DID's. I would have thought that the info on the 3CX site would have been enough to get them working as it is a popular feature. But you are right, using the DEMO code, you should have access to all the features of the paid version, with the two call limit.