similar to: Searching For a Voip Provider

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");

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

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: 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

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 want to connect a wi-fi phone to my Asterisk box through a wi-fi AP so I can make voip calls. please send me your recomendation about what wi-fi phone I should be looking for. Anybody tried the HOP1502 Wi-Fi IP phone. Its listed price $39. Regards; Chawki ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

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 installed G729 from Diguim and I was expecting the sound quality on my i686 machine to be better than gsm. Compared to gsm, G729 sounds closer and a little robotic. Is this what is supposed to be or am I missing something? I am interested in G729 because the internet in my country is very expensive and I want to save every bit possible. I want to use G729 because it takes less bandwidth for

Hi: I read many documents and I posted my question several times here without luck. I hope someone can help now please. Here is an example to demonstarte my problem: Suppose you manage the FWD server, how do you define an IAX client behind nat so he can receive calls from FWD. NAT client would register with FWD to let it know how to locate it. I just don't see how FWD finds the nat client.

There was errors when I tried to start the script recommended by Andrew to boost bandwidth for voip http://www.mixdown.ca/~andrew/dump/rc.tc. This is the output I get : ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File

Hi: I live in a country where calls from landline phone to a mobile phones is more expensive than mobile to mobile. I have FXO card connected to the landline. All the calls from IAX goes through this interface to the pstn and mobile phones. I want to save money by transfering mobile calls through a mobile phone. Is there some interface between the FXO card and the mobile phone so asterisk can

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

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

Hello everyone, I have an IAX server (asterisk@home) with a FXO card. I have a trunk connected to a voip provide, asteriskout. When I call my server from a public phone, I want to route this call to the asteriskOUT trunk so that I can make long distance calls. How can I setup a secret password in the extension.conf, so that my asterisk server can allow me to make long distance call ? Thanks

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Adrien Laurent > Sent: 02 August 2005 14:56 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] How to create a secret code to use > myasterisk@home server's long distance plan from a public phone > >

Hi All: Is there a script that I can download so I can make Asterisk send calls through a cellphone? Regards; Chawki Hammoud __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com

Hi: I have one TDM40B and one TDM04B on my Asterisk box. Both were working fine. Then, all of the FXS ports started to make echo sounds when I make FXS to IAX or SIP connection. All of the FXS ports fail to make bridging to the FXO channels. And when I try to make a call from the console to the FXO ports, I only hear static noise. Any suggestions about the source of the sudden change in

Great idea Dean, "I would also suggest that maybe instead of looking to set up within the disaster zones that you consider setting up in the relocation zone (eg where people are being sent to) yes they have payphones there but having a bank of 20 grandstream phones connected a T1 is a far smarter and more effective solution and probably more meaningful." > -----Original Message-----

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]: