similar to: max asterisk load

Hi folks, Is it possible (asterisk 1.6) to trigger the playback of an audio file in the middle of a call using the Manager Interface? I'm looking for something like AMI PlayDTMF command but for audio files. Thanks a lot, G. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Somebody has tested succesfully the speex echo cancellator? Sometimes ago I post the same question, and it was very untested. Best regards, and thank you very much for your work. G.

Hi, When i include a ',' in the application data, for example: exten => 1,1,Plaback(echo,file); [My file name is "echo,file"] the Playback application receives echo|file as data parameter. How can avoid asterisk change the ',' character? (The change code is in asterisk/pbx/pbx_config.c Ln 1150). I'm developping an application for asterisk and i need to pass

An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20040715/b3253d77/attachment.htm -------------- next part -------------- ? Hi, I have some queries: I am using speex in Windows platform. 1. I am having some problem with the speex,for input 8KHz,8bits/sample mono stream i am getting output as 8KHz, 16bits/sample mono stream. I would like to know

AFAIK, that's a common point for all AECs. But some of them solve the problem by resampling on of the end to keep it in sync with the other. On Tue, Jul 7, 2009 at 5:14 PM, ggb<ggb at tid.es> wrote: > Thank you John. > > On 07/06/2009 11:03 PM, John Ridges wrote: > > ly synchronized, and therefore the clock drift adds a non-linear > factor to the audio path. The AEC

Hello friends I've problem with my Centos 4.3. On start some information: 1. serwer HP DL 380 x Intel(R) Xeon(TM) CPU 3.40GHz 2. RAM 4GB 3. HDD storage FC 4. kernel 2.6.9-34.0.2.ELsmp #1 SMP This serwer is working with another one (that some configuration) in RHEL cluster (locking by DLM), as a file serwer with samba-3.0.22. Samba have about 100 simultaneus sessions. Problem is with load

Max TE410P card on an Asterisk Hello, Does anybody know the max number of TE410P/TE405P card we can put in an asterisk box? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040504/f220d3ae/attachment.htm

I see my own number(or remote called num) instead of caller id on incoming calls on my BT-102. but on Xlite everyything is OK. I'm using * latest CVS. - shabanip

The problem with different sound cards is that their clocks are not usually synchronized, and therefore the clock drift adds a non-linear factor to the audio path. The AEC can only cancel linear changes to the audio path, and so the AEC never converges.One solution is to measure the clock drift and resample either the input or output signal so that they *are* synchronized, and then the AEC

I get this message at CLI. what does it mean? - shabanip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040925/dbcd8d80/attachment.htm

hi, i'm under the impression that this feature is not available in asterisk, consider this scenario: - you are the operator. you answer a call from outside and you want to transfer it to one of the extensions. after you transfer, if the person you transferred the call to, doesn't pick up or if his line is busy, the call is transfered back to you, you can speak to the caller and tell him,

Where does the date/time stamp from Caller ID come from? On my extensions ATA188 and IAX2 soft phone the caller id date / time is 12/30 12:00AM. The Linux time is correct. SayUnixTime return the correct time. Any Ideas? Does this work? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I would like to test the echo cancel algorithm available in unstable version 1.1.4. This echo canceller can be used with other codecs like G711? Somebody could send me some documentation or sample, or explain the next functions: SpeexEchoState *speex_echo_state_init(int frame_size, int filter_length); void speex_echo_state_destroy(SpeexEchoState *st); void

Hey, I'm still waiting for the new file server hardware to arrive so I plan to enable user logins in the meantime without profiles and home directories. Reason for this is quite some pressure from people. The setup on the 2 DCs should be done a day or two. Could there be problems if I configure profiles and home directories some weeks after users logged in for the first time? What if e.g.

Hi all, I want to make the following: I want to run a linear regression on each column of a matrix "estima" on the correspondent column on the matrix "estima2". You see I want to regress estima[,1] on estima2[,1] this way to all columns.... At the same time I want to make a regression adding each time a new observation. You see, the first regression will regress only one

There is experimental echo cancellation code. Look in speex_echo.h and mdf.c. Unfortunately there's no sample code, and I don't think anyone's had much luck getting the echo cancellation to work at all. Would anyone like to prove me wrong? I'd love to see this work! I know it's really hard but just tonight the person I was talking to asked me, "Is there ever

Hi, I''m starting to use Test::Unit to create unit tests for my Rails Applications/Plugins. I don''t have much experience in Ruby Unit Testing. The problem is that the output report is very simple and it is very difficult to discover where is the error. Test::Unit does not show the line number of the error and the Stack Trace printed using the --trace option does not help either.

Hello, I am trying to setup a file server using the SFTP protocol with OpenSSH. I am in trouble because sshd refuses to chroot to a directory that is writable by users other than the owner. I guess that this is to prevent someone else from creating a .ssh/authorized_keys file and impersonate the user. But we have configured an alternative AuthorizedKeysFile. I also understand that a chroot user

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can i control not to run more than n simultaneus agi applications? Thanks in advance. rv -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140710/70b38e21/attachment.html>