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  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
  • Christopher Faust, Wed 07 of May, 2008 [15:28 UTC]: When I try to startx I ge input not supported. Though before installing asterisk I had no video issue to start the GUI
  • Christopher Faust, Wed 07 of May, 2008 [15:26 UTC]: Hi Nick, I got centos 5.1 and asterisk up But now I cannot start startx I have set the depth from 24 to 16 for the video i810 driver for the i845 on my netvista machine but I cannot start GNOME. Please advise
  • Nick Barnes, Wed 07 of May, 2008 [10:01 UTC]: Howard - You'll need to provide a lot more information if you really want help.
  • Nick Barnes, Wed 07 of May, 2008 [10:00 UTC]: Christopher - Search the Wiki and you'll find a page I wrote detailing exactly what you have to do for Asterisk 1.4 + CentOS 5.1.
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Should you have any problems with this service, please let me know.

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Created by jht2, Last modification by limiao on Mon 14 of Apr, 2008 [05:42 UTC]

Comments Filter

Re:Vip Fone To Call Free JUst By watching vedio

by Valter James on Wednesday 06 of February, 2008 [06:30:48 UTC]
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@ rupesh

by GehirnJogging on Friday 25 of January, 2008 [17:41:48 UTC]
yes, where is the point?

Re: Vip Fone To Call Free JUst By watching vedio

by rupesh on Friday 04 of January, 2008 [08:52:39 UTC]

Re: Vip Fone To Call Free JUst By watching vedio

by rupesh on Friday 04 of January, 2008 [08:51:34 UTC]

Re: Vip Fone To Call Free JUst By watching vedio

by rupesh on Friday 04 of January, 2008 [08:49:02 UTC]

Re: Vip Fone To Call Free JUst By watching vedio

by rupesh on Friday 04 of January, 2008 [08:45:20 UTC]

VoIP Security Solutions

by jenniferhan on Thursday 27 of December, 2007 [06:35:09 UTC]
SpeedVoIP is a professional VoIP Security and VoIP anti blocking solutions provider.
The core solution for VoIP Security and VoIP anti-blocking is VGCP (VoiceGuard Control Protocol).
It can work with any 3rd-party Softphone / ATA / Gateway / IP Phone / IADs and SIP proxy or server.
It can work in the way similar to that of SOHO router, but it only encrypts and decrypts SIP and RTP packets on link layer, not to handup these packets to IP stack for forwarding while bypassing other data packets originating from SIP terminals. In this scenario, peak throughput and minimal CPU overhead can be easily achieved.

VoiceGuard can real-time incorporate light-weight traffic for puzzling and bypassing VoIP blocking system without consuming more bandwidth and compromising voice quality. Even in some circumstance, VoiceGuard can simulate traffic behavior of universal data networking protocol such as OICQ, MSN and so on.

For more information, please refer to: http://www.speed-voip.com/index-36.html

Andy
xd.wong@speed-voip.com
andywong-01@hotmail.com

Please help !!!

by Franklin on Sunday 12 of August, 2007 [01:31:29 UTC]

Hi. I'm relative new to all this. I'm planning to install my current PBX in my office. I'll start first using WindowsXPHome and in a near future change it to a Linux system.

In my office we have 2 phone lines that our provider give us regular RJ11 jacks also we have another 2 lines that current provider use something similar to the:
Grandstream HandyTone 286 SIP Gateway ATA HT286, so the 2 lines are each one in 1 device of those.
In total we have 4 lines. But keep in mind I plan to grow in lines.

Since I'm new in all this I've some questions:

1.- What hardware does I need to connect the 4 lines (2 regular and 2 that comes in ATA) to the asterisk? I supposed I only need a device for the curren 2 regular lines, think that the ones in ATA doesn't need any hardware. But not sure what I need?

2.- For what did you use or recommend using each of this devices:
ATA
Multi External Device
PCI card

3.- Why some ATA mention they are Unlock? Is this only applied to connect it to a VoIp provider?

4.- Which are the differences between FXS y FXO, what are they for?

5.- What do I need to know about SIP and IAX2 protocols?

6.- What does I need to use any standard Phone, instead of softphones and costy
IpPhones? Do I need extra ATAs? or something like that? If so, how many I need? 1 for each extension where I plug a regular phone?

7.- If I use Asterisk to do the following do I need to consider any special hardware? If any of this features is not possible please tell me that to stop dreamin on it:
7.1.- Get into any computer outside my office and with a SoftPhone conect to my office usin ALL the features as if I were there WITHOUT limits.

7.2.- What hardware do I need to use an standard phone in my home, but connected to my office, as I were there, without the need of a computer?

7.3.- I use a PocketPC with version Mobile 2003. Someone told me that something could be done to use my device as something like a SoftPhone and use all the features as I were on my office. What do I need to use this?

7.4.- As I know all of this features are included with Asterisk: Menu Attendant, voice mail that could be read from anywhere, voice mail delivered to email, MP3 music while onhold, CallRecording. All of them are truly included? Do I need extra hardware for any of them?

Thank you for your help. Write to me at fcanavati@gmail.com

ARI

by Shaman King on Monday 02 of July, 2007 [09:12:30 UTC]
Hello, I encountered error in the ARI Recordings. It says
Path is not a directory: /var/spool/asterisk/voicemail/default/102

Re: Listening to a conversation in progress...

by Steve Vick on Wednesday 21 of March, 2007 [19:15:16 UTC]
Not if he wants to break the law. It's against the law in the US to listen to a conversation without giving explicit or implicit warning.


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