All, I was wondering if anyones tried to use OR currently use the Microsoft Speech Server or their UCMA 3.x SDK etc. as their ASR/TTS backend/engines etc. If yes, then whats their experience? Please Note, this does NOT need to be integrated with Aster..
all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. ..
I thought one possible solution might be creating an [applicationmap] that essentially handles the assisted transfer manually. Ive done a great deal of reading on this matter and aside from the fact that Im still a bit fogy as to how i would e..
Users, Im planing to implement call completion feature in asterisk 1.8 but having some issue. I am following this document https://wiki.asterisk.org/wiki/display/AST/Generic+Call+Completion+Example I am getting error non-zero error on console. I am us..
Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 g..
Are sip.confs defaultexpiry and maxexpiry global?
Or can they be used on a per-extension basis?
Id like to force some extensions to re-register more frequently than others (server-side).
hi i searched a lot but i couldnt find the answer i have two openvox(fxo/fxs) card so I have 24 ports! on first card i have 12 fxs and on the second i have 12 fxo i want to then one person calling fromdahdi/13 forward it to dahdi/1 when a person call..
All, First post here. I am dialing out via call file to remote number, when call is connected a local number is dialed. And on success both calls get bridged and works fine. This is a parallel auto dialout application. I want to set a variable as s..
Just verified I faced the same issue once and got it reolved by adding /n like
Channel: [mailto:Local/0036701234567@CustomCallOut-1/n] Local/0036701234567@CustomCallOut-1/n in you ca..
Hi Were getting requests coming in for higher quality audio in our call recordings. We currently use MixMonitor and everything is being saved in its native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a convers..