I followed the instructions on this site for setting up Follow-Me and VMx Locator to work together. It appears to be working fine, well, trying to. I only have one Zap channel and one Sip account. When a call comes in the Zap channel (my main business line) and the caller presses ‘1’, it tries to dial out to my cell phone but fails since the Zap channel is already in use with the incomming call. How do I force it to dial out the Sip line??
you either fix your normal route to have it automatically fail over to the SIP trunk when the ZAP trunk is occupied, you setup a new route that requires a prefix to force it into calling out a new route that is connected to the sip trunk, or you dig into the VmX and manually update the astdb structure used to control it along with possible custom written context. The last alternative is completely un-necessary though unless you were doing something very unique.
How do I set the Zap trunk to fail over to the Sip trunk? I am using FreePBX 2.4 and I defined an outbound route to try ZAP/3 first and then SIP next. Or at least I think that is what I did. On the outbound route screen in the trunk sequence portion I listed the Zap first and then the Sip trunk. But the VMx Locator agi scripts must not know anything about that. When I watch the call process via the CLI, I see it try the ZAP channel and say UNAVAILABLE and drop right into voicemail. It never tries the SIP channel.
The VmX should simply make a call down the Local Channel just as if it were dialed from a phone. From what you described, it should be going out the second trunk when the Zap trunk is unavailable. Basically, if you can receive and inbound call and then make a separate outbound call, that would mimic the VmX. There are no AGI scripts involved. It simply jumps into from-internal with your number unless you have tweaked with some of the more advanced parameters.
Hmm, not sure what is happening then. Watching the CLI I can see it launch the dialparties.agi script get launched. I have not taken the time to browse through the script to see what it does tho. I did verify there is such a script in the /var/lib/asterisk/agi-bin folder.
I just picked up two phones and dialed out on them. The first one went out the Zap channel and the second one went out the Sip channel as it should. That part seems to be working. I just can not get it to do that via the Follow-Me / VMx Locater methods.
try posting a trace, maybe someone can spot it for you. And make sure to specify how you have your VmX configured.