similar to: sound-recorder crash when I start Asterisk

Hi every one: I appreciate the conrtibution every one is making and please forgive me for my question. I have Asterisk running on Linux Redhat9 dstr. I subscribed to a third party sip providers to make LD calls. Can I initiate a call sessions from asterisk CLI> command prompt after I configure extensions.conf and iax.conf? In case I need a soft phone, what is the necessary configuration that

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

I appreciate any recomendation of a simple CallingCard Application and resources of users manual. __________________________________ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/

Hi: I unseccessfuly tried to place a call from one IAX behind a nat through another IAX with a real IP. I got the following error message on Asterisk Console: Executing Dial("OSS/dsp", "IAX2/wassim/01)") in new stack -- Called wassim/01) Call rejected by 195.112.214.98: No such context/extensionad: -- IAX2/wassim-1 is circuit-busy Aug 8 10:13:38 NOTICE[21759]:

Is there a way to hide the callerid on analog line on outgoing calls. Any ideas whether it could be done through configuration, a script or hardware. Thanks; ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

Hi all, I have deploy speex 1.1.6 in my application. With no option set, I can hear that the voice sounds a little bit artificial like robot. Any idea what causes this? I use openh323 with speex, but it seems ok. Is it neccessary for me to use more other filter prior to encode the sound or after decode my sound? yueweng __________________________________ Do you Yahoo!? Yahoo! Mail -

Hi: Please advice me of a voip provider with reasonable reseller program. I was using voipjet and it has a lot of problems. Did anyone experienced asteriskout.com service? They have good prices. Regards; Chawki Hammoud ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele

hi, im a student from a local university in our place. i'm not sure if this is the place to post this question so please pardon me if im wrong. my questions is that when i transfer speex encoded samples through a rtp session wire, upon receiving the samples i immediately decode and play them, but i get a crackling sound when i decode and play each sample received. 1) i used 160*5 as my wave

Hi, This applies to everyone having (or suspecting) problems with Speex. The first thing to do is to encode the file in wav format and use speexenc/speexdec on it. If you're getting something different with your application, it's likely buggy. If the result isn't OK, then it can be: 1) Normal given the bandwidth/bit-rate used 2) A conditioning problem with your audio (i.e. DC not

This is my second attempt trying to get help and I am hoping someone can. When the musiconhold extension is matched, Asterisk attempts to execute musiconhold and stops right away, this is what I gets: Executing MusicOnHold("OSS/dsp", "") in new stack -- Started music on hold, class 'default', on OSS/dsp -- Stopped music on hold on OSS/dsp Is there a file that

Hi: I had an fxo card from Digitnetworks and it was working fine on my Asterisk box. I then replaced it with TDM04B. I changed the zaptel and zapata to include the four channels. When I run ztcfg, I get configuration errors: Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04:

Hi: I use shared internet bandwidth and the calls are very clear from around midnight till about 4 pm when it goes bad after that. Is there a way to boost the internet bandwidth for Asterisk at the peak time? Thanks Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html

--- Mike Mestnik <cheako911@yahoo.com> wrote: > Date: Fri, 25 Jun 2004 09:51:21 -0700 (PDT) > From: Mike Mestnik <cheako911@yahoo.com> > Subject: Re: IPP2P: Simular project l7-filter. > To: Eicke Friedrich <tady@gmx.net> > > --- Eicke Friedrich <tady@gmx.net> wrote: > > Mike Mestnik wrote: > > > http://sourceforge.net/projects/l7-filter/

Good day sir and Thank you Luke for replying. Whatever U told about the things like accounts on linux and windows machine and adding a user to smbpasswd with -a option but still it is not woriking. Please help me . Thanks in advance. --- Luke Roberts <Luke.Roberts@isispc.com.au> wrote: > Hi Prakesh, > > Do you have linux user accounts to match your > windows

ah. I was only talking about removing support in the winamp2 and xmms plugins. the decoder libraries will still be able to skip over/ignore id3 tags as before. I think AudioReQuest and PhatNoise must be using their own code to read id3 tags, which won't be affected. Josh --- Steve Vasquez <steve.vasquez@request.com> wrote: > Hey Josh, > > We currently use id3 v1/2 in the

Hi: The phone company here use Europe telephone system. They offere ISDN lines with multiple channels, starting with two and they add channels on demand. Am I write to understand ISDN of more like E1 or T1 line. If not, what's the difference. __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

I have gnome-sound-recorder on my FC10 system, but cannot find it for Centos. Is there a way to get it for Centos or a similar package?

I bought a Cisco 7940, I need to configure it for Asterisk. I checked the wiki pages. Followed the link to Cisco web page. Tried to download the image for SIP. It wo'nt allow me even though I registered for the CCO Valet. Is the image available anywhere else? I saw some of the messages in the mailing list that it supposed to be fairly simple. I would highly appreciate if somone could post

When I load the latest driver from Nvidia in CentOS-3 the logon screen turn light gray and I can't log in. Has anyone else had this problem? If so what did you do about it. If I use the driver that comes with CentOS I don't have any problems. __________________________________ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail