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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • 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.
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FXS-FXO Converters

If you want to create a VOIP to PSTN gateway and only have an
ATA with an FXS port, you can use an FXS-FXO converter connect the FXS port to a PSTN phone line.

    Internet------ATA-----FXS-FX0 converter------PSTN phone line

This has been a popular option for creating a low cost PSTN gateway because of the lack of low cost ATAs with a FXO port. This option will probably continue to be popular even with the advent of the Sipura 3000 given the increasing numbers of VoIP service providers who provide users with their own nominated preconfigured brand of FXS gateway or ATA.

If you are using an FXS-FXO converter with a Sipura ATA then see this
mailing list posting.

Sources of FXS-FXO converters
  • Auerswald A-Box
  • BroadTel FXS to FXO Port Converters: 1 to 1 mapping or 1 to 2 mapping - and many others.
  • ICS France FXS to FXO Port Converter . 4-48 port FXS/FXO Gateway,
  • PCPhoneline.com - VPC1000 FXS to FXO Port Converter - Only $39
  • SIPCPE NEW FX-300 Analog Call Director, a converter and 2 line switch in one unit with auto busy tone detection, volume boost, echo cancellation and CID authentication. sales@sipcpe.com SIPCPE also has a new Cellular (GSM) to Analog or VOIP converter/switch called the FX-300 GSM Call Director.
  • Welltech 2/4-port FXS, 2/4/6-port FXO, 1/2 FXS+FXO, IAD 161/162, IP phone
  • PBXEQ
  • X100P.com announces STOC-FX, Professional Grade FXS to FXO Converter for All VoIP ATAs and Asterisk FXS Ports
  • YUXIN 1FXS ATA,1FXS,1FXO ATA,2FXS gateway,4FXS gateway,FXO IP Phone
Created by jht2, Last modification by ynccarol on Mon 12 of May, 2008 [02:44 UTC]

Comments

Comments Filter

Adapter with FXO-FXS

by Adam King on Thursday 24 of April, 2008 [10:28:16 UTC]
JR-901+ is an internet based voice network phone terminal. PHONE series IP phone adopts multiple voice control protocols and voice compression coding methods to directly convert analog voice into IP packet for internet transport, thus effectively using the existing bandwidth to provide PSTN quality voice service.


Support SIP and dual public server
Support Support Bridge and Router model .
Support basic NAT and NAPT
Support DHCP Client on WAN .
Support DNS Relay, SNTP Client, Firewall .
Support VPN and VLAN


We are one of the leading manufactures of VoIP (Voice Over IP) terminal products in China, we dedicate in designing, producing top-ranking IP phones and gateways, and all products are CE, FCC certificated. Ascribed to our excellent quality control and competitive price, our products have won a good reputation both at home and abroad.

Please visit our website: www.iplink.cn , we hope you will find some items interesting. We shall be pleased to provide you our best prices. Also we could provide the OEM/ODM service. Please don't hesitate to contact me for further information.

Adam

JR Information Technologies Co.,LTD.

E-mail: iplink.adam@gmail.com

       adam_king@iplink.cn 

Skype: adam_king88

Tel: +86-755-89742562

Fax: +86-755-89742889

Mobile: +86-13424286541

MSN: strongpropeller@hotmail.com

Address: East, Floor 1, periods 1, Xintianxia Industrial Park,

Banxuegang Industrial Area, Longgang District,

Shenzhen 518112, China

www.iplink.cn

Re: MODEM TAPI IN ASTERISK

by Kevin Komara on Saturday 22 of September, 2007 [07:19:23 UTC]
Did you actually get the TAPI modem to work as an FXO ? I got my partially working. I was able to initiate a SIP to PSTN call. The analog phone rang and I was able to answer it. I could hear my voice on the SIP softpone when I talked into the analog hand set, but could not hear my voice on the handset when I talked into the mic on my PC. I posted numerous questions on the Asterisk win32 forum, and was told the problem is my modem is only half duplex and I need a full duplex voice modem to work. So far nobody has replied with a working full duplex modem. Did yours work full duplex ? If it did please let all of us know which modem you have !!!
Thanks in advance,
Kevin K.

MODEM TAPI IN ASTERISK

by Rodrigo on Tuesday 19 of June, 2007 [11:49:03 UTC]
Hello i downloaded asteriskwin32 (for windows) and the lastest version works with tapi modems (voice modems), you can use it as a fxo card. I would like to know if that is possible in the linux version (i´m working in suse 10.2), thanks in advance

SIP CPE SOURCES SIPCPE.COM

by cbolton6001 on Friday 30 of June, 2006 [16:24:49 UTC]
SIPCPE / sipcpe.com is one of the worst suppliers of the FXS to FXO Port Converter (FX-200)! SIPCPE does not know the product. When they claim that they developed and wrote the specs for the product. SIPCPE is extremely difficult to deal with, along with poor customer service. I would highly recommend all other distributors, for which they are knowledgeable and courteous.

how to connect asterisk with exist PBX

by wichaya sropas on Tuesday 21 of March, 2006 [15:30:09 UTC]
I have Avaya and Nortel. How can I connect it with Asterisk? Can any one help.

how to connect asterisk with exist PBX

by wichaya sropas on Monday 20 of March, 2006 [14:54:23 UTC]
I have Avaya and Nortel. How can I connect it with Asterisk? Can any one help.

PCPhoneline

by razametal on Wednesday 15 of June, 2005 [05:12:17 UTC]
I´ve a device from pcphoneline but i can´t make it work.. any 1 haves a instruction manual?

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