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Pass DID to SIP tie trunk

I would suggest you might add exten => s,1,dumpchan() before ShoCanTel: exten => s,1,Macro(dialout-trunk,4,${FROM_DID},,)

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Unable to check voice mail from outside line

When and if you set up the voicemail box the greeting will be as long as you make it. All sound files are below /var/lib/asterisk/sounds , the text of each of them is listed at...

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Pass DID to SIP tie trunk

dicko: exten => s,1,dumpchan() I tried this and just got all circuits are busy. lgaetz: FreePBX version? (padding) 2.10. Old, I know. It's on purpose. I tried the latest version and it did not play...

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Pass DID to SIP tie trunk

did you increment s,1 and s,2 to s,2 and s,3 in your macro ? Use s,1 then s,n . . . . Did you you look at your logfiles/console?

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Pass DID to SIP tie trunk

Hi! ShoCanTel: I tried the latest version and it did not play nicely with my SIP provider. You mentioned Vitelity as being your provider in another thread and there are people using that provider with...

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Pass DID to SIP tie trunk

ShoCanTel: exten => s,1,dumpchan() I did forget to increment. Fixed it, calls now route, but still sending "s" as the DID. [2015-12-19 15:28:36] VERBOSE[1931] pbx.c: -- Executing...

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Pass DID to SIP tie trunk

So. . . what did the dumpchan() function call produce in your logfile/console output ?

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Pass DID to SIP tie trunk

Weird.....the DNID digits are all correct, but I just don't get why it's then setting the DID as "s" Info:Name= SIP/vite-inbound-0000000fType= SIPUniqueID= 1450561054.15LinkedID=...

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Pass DID to SIP tie trunk

Then use the CallerIDNum= 314XXXXXXX if 314XXXXXXX makes sense, to set the CallerID(num) before bridging to the tie-line, also IMHO macros should be replaced with gosubs or just simple gotos if...

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Pass DID to SIP tie trunk

I'm sorry but I don't quite follow. I'm not so much concerned about Caller ID as I am what number the caller dialed. Both the called number and the number I called from are in the 314 area code, but...

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Pass DID to SIP tie trunk

Then you will need to specify that particular DID as a landing point for your inbound calls or the catchall inbound route will use the "start" extension , namely s. Call it entropy at work

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Pass DID to SIP tie trunk

So literally just create an inbound route for each individual DID, instead of any DID?

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Pass DID to SIP tie trunk

Holy crap that worked. I've spent hours on this! Thans for your help!

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Pass DID to SIP tie trunk

Did you read the Wiki yet?

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Pass DID to SIP tie trunk

No comment.....

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IVR Custom Recordings and DTMF

The DTMF (dual tone, multi frequency) detection process uses "Fast Fourier Transforms" to isolate the [4x4] matrix DTMF tones from the audio stream, not volume or anything else, if your recording...

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Poll: FreePBX Phone Apps - What brand should we add next?

My vote is for Polycom. I have several clients that run Polycom phones that would definitely purchase the Phone Apps if Polycom was supported.

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Ajax request declined

I have the same problem on a new install of FreePBX. I have created an bug report.

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FreePBX 13 language setting for PJSIP trunk

Hi, How setup language for PJSIP trunk? FreePBX 13.0.35 - Asterisk 13.6.0

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Upgrading to PHP 5.5> on FreePBX Distro 13

Half a year ago I tried to upgrade my FreePBX 11 machine from PHP 5.3 to 5.5 and it broke the GUI altogether. Probably due to some proprietary rights files you've installed. I am now with 12 version...

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