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 4.5 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 4.5 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 4.5 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 4.5 years ago 4 Answers