similar to: VOIP CONNECTION, NO AUDIO AT THE OTHER END, NEWBIE

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: 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]:

i installed Asterisk on linux FC3 box and i was able to make third party calls from voipjet. I then installed an X100P from digitnetworks and i was able to execute modprobe zaptel and modprobe wcfxo and i had to add some lines to the file: 50-udev.rules before i was able to perform ztcfg without errors. I had no problems running Asterisk before i installed X100P, but this is not the case now.

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/

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

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

--- 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

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: 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

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

Hi, I'm a newbee to all this asterisk stuff, and after reading a fair amount of docs at the voip-info wiki, I was wondering if it will be possible to work out something as the following scheme: A B +-----+ +--------+ +----------+ +---------+ | SIP | | | Internet | PC | phone line | classic | |

Here's another one for ya, I can ssh into my server from workstationB but when I try to ssh into workstationB from my server I get this error message. @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ @ WARNING: REMOTE HOST IDENTIFICATION HAS CHANGED! @ @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ IT IS POSSIBLE THAT SOMEONE IS DOING SOMETHING NASTY! Someone

/usr/local/sbin/smbldap-useradd -w computername Error looking for next uid at /usr/local/sbin///smbldap_tools.pm line 879, <DATA> line 283. I cannot add machines to the samba pdc because of this... everything else with samba/ldap seems to work, I have the feeling it is related to this # Where are stored Idmap entries (used if samba is a domain member server) # Ex

--- Dan Sully <daniel@electricrain.com> wrote: > I'm attempting to use the metadata level 2 interface to get at the > various > interesting bits, which is working fine. But I also need a relevant > piece of > data, the offset of where the metadata ends, and the audio blocks > begin. I'm > not finding a clear way to reach that number. Is this exposed >

sorry about the delay... this it the first I'm seeing the message, maybe it got stuck in moderation because of the cross posting. --- Christian HJ Wiesner <chris@matroska.org> wrote > I was wondering how suitable both are with respect to video editing, > especially if it comes to sample precise cutting. What is the > absolute minimal number of samples per frame in a FLAC

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 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

Sorry for the very newbie-like question. I have the FXS part straight. The part I don't understand is the FX0 part. Will I need the FX0 card if I am connecting to a service like FWD? My goal is to get rid of my phone line all together. I am under the impression I will only need an FX0 if I'm connecting to the central office side of the phone connection or to an existing PBX. Bottom