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...
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...
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...
Johnthephoneguy,
I have plans to try and make sure that it doesn't. I can tell you there is already a mirror running, but I don't want it to go that way, I want his site to stay up.
-Drew
I'm unsure what version our polycom phones are on but it could have to do with the digit mapping blocking the 9. I know we had to tweak that around in the provisioning files to make it work with my Asterisk system for use with some extension patterns.
I was notified this morning by our leadership at work that Gene Trouteaud, who many of you know here by GHTrout, passed away unexpectedly this morning. I have been lucky enough to have been working with Gene for the last several months but his work here as well as his own site has helped me out...
I have set up a TIE line between my CS1000M and a brooktrout fax card that belongs to a fax server.
I have set up DSC's to allow certain extensions to be routed over that TIE line but I need for the fax server to be able to make use of the PSTN connections on my Nortel for outbound faxes. I'm...
I have no idea how it became active and busied out the trunks, but after giving the consoles back control to manipulate it (ITG) which I had taken away trying to fix an issue that I dind't understand at the time) I was able to press shift + the key and after that the console stopped lighting...
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