Microsoft Speech Server/UCMA Integration

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Hello All, I was wondering if anyone's 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 what's their experience? Please Note, this does NOT need
to be integrated with Asterisk ala MRCP or some module/plugin etc. I just
wanted to know if someone's used it and and what their experience has been
in both, TTS and ASR. Thanks
RR

Asterisk Users 4.6 years ago 0 Answers

echo when calling to the pstn

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Hello 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. Can you help me with this issue? Best regards,
Vitor Flausino

Asterisk Users 4.6 years ago 0 Answers

Manual Call Transfer (Perl, Asterisk::AGI, MySQL)

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Hello Everyone! I've hit a bit of a roadblock and I am hoping that someone might point me in the right direction. I am using Asterisk 1.2.4 - I do not have the option of updating it, please do not waste your time telling me to =) - I am using PERL AGI scripts to maintain an "active calls count" field for each phone in a mysql database table, for example (not actual code, just trying to illustrate) $SIG{HUP} = 'IGNORE'; .... mysql_update_call_count($user_id, ($count +1) ); $dialret = $agi->exec('Dial', $dialstring); mysql_update_call_count($user_id, ($count -1) ); This works great, except when doing assisted transfers (or any transfer for…

Asterisk Users 4.6 years ago 0 Answers

fail-over server

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Hi, 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 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to server2 and that server2 knows that they are actually "on-line" so calls can be routed to them. How can I minimize this time lapse? Can Asterisk "notify" all SIP clients…

Asterisk Users 4.6 years ago 4 Answers

SIP registration

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Hi, Are sip.conf's defaultexpiry and maxexpiry global?
Or can they be used on a per-extension basis? I'd like to "force" some extensions to re-register more frequently than others (server-side). Thanks, Vieri

Asterisk Users 4.6 years ago 3 Answers

forward calls by the ports

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hi
i searched a lot but i couldn't 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 from dahdi/13 forward it to dahdi/1
when a person calling from dahdi/14 forward it to dahdi/2
when a person calling from dahdi/15 forward it to dahdi/3
........
how can i do this?
i should make an AGI? or can i make it with extentions.conf? how can i get

Asterisk Users 4.6 years ago 0 Answers

Set variable on Call Answer

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Hi 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
soon as the local number answers the call, so that system won't try to
dialout that local number again and stops further dialing. What should be
the best way to deal this situation ? Any help would be appreciated. Thanks
-dani

Asterisk Users 4.6 years ago 1 Answer

Call files error

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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 case.

Asterisk Users 4.6 years ago 1 Answer

Call Recording audio file quality query

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Hi We're getting requests coming in for higher quality audio in our call
recordings. We currently use MixMonitor and everything is being saved in
it's native 8000Hz, 16 bit wav format. I have seen information on using Monitor and specifying a conversion to
mp3 when the call ends and the 2 channels get mixed but surely the 2
channels are already saved as 16bit 8000Hz wav files so the quality is
lost already? Is there any way of making high quality recordings of call content? We're currently using asterisk 1.4 and soon upgrading to 1.8…

Asterisk Users 4.6 years ago 5 Answers