* You are viewing the archive for February 8th, 2011

Microsoft Speech Server/UCMA Integration

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.


echo when calling to the pstn

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

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

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)

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 that matter).

We have Polycom IP550 Phones which can do the transfer with a button, As an example of this process and the problem, and assuming these are all internal phones dialing extensions…

phone A dials phone B
phone B presses transfer to transfer phone A to phone C
phone B hangs up

Because the Dial command in the AGI script executed when phone A called phone B is still running the active call count remains at 1 for phone B until the call between A and C ends (at which point they all zero out).

I also tried using atxfer to resolve this problem and got a different behavior phone A dials phone B

phone B presses *2 then phone C’s extension to transfer phone A to phone C
phone B hangs up

an active call count remains at 1 for B and C but A drops to 0 count.

Might be worth mentioning the possibility that phone B is already on the line when the call from phone A comes in.

I thought one possible solution might be creating an [applicationmap] that essentially handles the assisted transfer manually. I’ve done a great deal of reading on this matter and aside from the fact that I’m still a bit fogy as to how i would even do that,.. it seems that there is still no way for me to determine who is being transferred when the second channel is opened (new uniqueid / agi script execution).

Is there perhaps something I am missing which would help resolve this?

I hope that I’ve explained my problem clearly. I have only been tinkering with asterisk for about a week so I apologize if I’m not using the appropriate vernacular.

Asterisk CallCompletion dialplan

Hi Users,

I’m 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 using softphone x-lite

root@tux:/etc/asterisk# asterisk -r
Verbosity is at least 3
== Using SIP RTP CoS mark 5

fail-over server


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 in its sip.conf that they need to acknowledge being on-line or not (thus forcing re-registration in my scenario)?



SIP registration


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).