I have a dk424 connected to an Asterisk call manager via PRI T1. I would like to replace the existing voicemail (which is now forwarding to 1 of 8 fxo ports) by forwarding calls out PRI channels and let asterisk handle the voicemail.
Perhaps I should ask this question first, I am currently calling remote extensions (those accessable by asterisk) by dialing their 4 digit extension proceeded by a trunk access code (9). Can I set up the remote extensions so no trunk access code is needed?
Thanks,
John
Maybe there is an Asterisk guru on this board, but an Asterisk help group would be more likely to give you an answer as this is really about routing, not voicemail.
Did I say fxo - it should have been fxs.
What I wanted to do was re-direct calls destine to voicemail back out the PRI. We do this with merlin-legends and it works good. I assumed this would be no problem on the dk - but you never know. I'm not bad with Asterisk, so I can generaly get this to do what I want it to.
No Asterisk gurus have responded, you might repost with: Redirecting calls on a 424 - Asterisk involved
That might get more people looking at the post.
Should work for fxs but not for fxo. Does the Asterick know the vm id codes and integration patterns from the DK? The dial 9 will have to stay unless you want to put a button on the phones (like a pool button) to access the PRI directly.
I am retiring the T in favor of the PRI. I had the T's channels assigned as wink start tie lines. So I never really considered using it for station ports.
Can I re-define the RDTU2 as fxs ports (analog stations like RSTU2)? In that case I can just route vm back in to asterisk through that T (more or less just as my current vm channels work - and I can skip going back to the analog realm). Figuring out the vm codes should be straight-forward.
I object to the term asterisk-guru as it already has an amazing array of features. If I take a month off reading the development-posts (and who can read all that crap) capabilities go in that I never considered - so you just can't know it all. But I will identify myself as someone pretty proficient with asterisk. I recommend getting involved with it, it's the future...
You cannot define the RDTU as fxs ports. Are you using something like an Adtran TSU120 to do this?
What I meant was can I plug the RDTU's T into a fxs channel bank in order to have 24 additional analog stations?
I don't need something like an Adtran. I'm using a Digium quad-port-T1-interface in the Asterisk PC - and only using 2 ports, so I can configure the unused ports however I like so it would be nice if I could still utilize the RDTU (which is no longer used) for voicemail.
I'm not sure if I have stated the question clearly, I know I could host my voicemail like this (using my current toshiba configuration)...
8-RSTU2-ports-------8pairs------channelbank---asterisk
...but it would be better to do...
RDTU---T1---asterisk
Is there a way I could forward voicemail to asterisk with the 2nd setup?
DER? The RSTU ports are ANALOG single line ports.
The RDTU is CHANNELS coming or going, not analog ports.
I suppose you could have the RSTU ports access the RDTU to make an outgoing call and connect the RDTU to the Asterick through a T-1 card, but then I expect you could deliver most of the mail in the US if you had hundreds of thousands of horses and Pony Express Riders, but is that necessary or important?
Not a possible to do because only RSTU ports can be vm ports in program 31. You cannot specify trunk ports in program 31.
I meant a T-1 card in the Asterick, I didn't make that as clear as I could have.
RSTU to RDTU to T-1 card in Asterick. Even if possible, I fail to see the advantage.
The advantage that the second setup has over the first is that it wouldn't tie up an extra channel bank and consume additional power. Another factor is the extra digital to analog & analog to digital conversion. When working with VoIP (or cell phones) extra conversions should be avoided whenever possible. Additionally, it wouldn't cost me anything to deploy as we have both a RDTU and a extra asterisk T1 port that is not being used.
What implied I could do this was this line from section 6-2 regarding setting up the RDTU in the installation & maintenance manual:
'Note Each Tie or DID line decreases the system's station port and CO line capacity by one'
...I took that to mean I could address the tie lines (RDTU channels) as station ports. Unfortunately, it seems I can't.
An aside, we are using *40 (maybe in conjuction with something else I can't find at the moment - rather than 31) for voicemail routing.
Thanks,
John