Recent content by ddickenson

The bit in front of the @ symbol is just if you want it to "prepend" the mailbox you're trying to access. If you're trying to do something like a generic "everyone's" access where they are prompted to input their mailbox then it'd be something like this: exten =>...

lexer, I believe you are correct. I don't personally use freepbx gui on any of my systems (which I think is what elastix uses) but I recall when playing with it early on in trixbox distro that you couldn't make any changes to the actual extensions.conf file because of what you are describing...

AvayaCiscoAdmin, Did you ever get any resolution on this? I know you don't need a domain to run a sip trunk between avaya and asterisk. Also the VM number would get defined in extensions.conf. Something like this: exten => 1234,1,VoicemailMain(@default) If you leave it just (@default) then...

Have a good holiday! www.ghtrout.net

Alright, I've found how to find the info we need. Go ahead and make a call that will fail so it'll be fresh in the logs. Then in the freepbx webgui click on "Reports" at the top and scroll down to "Asterisk Log Files" You should see the "File" box at top says "full" Highlight and select all...

Depending on your current log settings we might be able to just pull historical data if they are verbose enough to tell what happened. Will respond again in a bit when I get the system up and familiarize myself with the GUI. Could walk you through the command line version easily but I'm not yet...

Ok, can you post one of the endpoints you're trying to call to as well. not being able to take a look at the system myself we'll have to figure a way to get some CLI data of a call that failed to try and figure out what's amiss. I'm installing AsteriskNOW in vmware as I type this so hopefully...

eddiebullman, it sounds like maybe there isn't an extension (or pattern match) setup to catch calls in the context that you have your sip trunk set in. Do you know how many digits your LG system is sending across the trunk? if 4 it could be as simple as something like exten =>...

Good deal. Glad you were able to make it work. I will say that I don't see a queue in there anywhere or for that matter anything that dials a phone. I'm assuming it's just not been added to the example. Sorry for the late response, the site normally tells me when something I've commented on...

emartins, Happy to help but will need to see some config files in order to go any further. Are you saying you call the analog phone line and it rings for a certain number of rings prior to ringing a sip phone and you want them to both ring at the same time? www.ghtrout.net

eddy123, I can help with this. Did you ever figure it out? I'll assume that the telco is giving you say 4 digits for your DID. I'll further assume that those 4 digits are 1234 just for example sake. I'll also assume that the context in which you have the working queue at an extension of...

ghtrout.com has been down for a couple of months, as noted here: http://www.tek-tips.com/viewthread.cfm?qid=1769054 We have been working to keep the actual ghtrout.com domain up but it's taking longer than expected for the transfer and such. I wanted to let the board at large who may not have...

As some of you may have noticed ghtrout.com is down. I've had a backup mirror running since Gene's passing and hoped we wouldn't have to use it as part of the transition for his site but in case someone needs something from it in the mean time here's the link: [link...

Also this may be of use depending on the release of your system: http://ghtrout.com/files/911_misdial_prevention.html

Here's how I did it. Here is an example of ESA – this is what we use. This is how to get 911 and 9,911 to work – and get misdial prevention >ld 24 REQ prt TYPE esa CUST 0 CUST 0 ENTR 0 ESDN 911 ESRT 29 DDGT 911 MISDIAL_PREVENTION YES MISDIAL_DELAY 4 ALOW_LASTDIG_REPEAT...