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Sat 17 of May, 2008 [07:17 UTC]

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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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Soundwin Network

http://www.soundwin.com

Soundwin is a VoIP Equioment manufacturer
VoIP Feature:
  • Caller ID Delivery and Detection: FXS support DTMF&FSK Caller ID generation; FXO
supports DTMF&FSK Caller ID detection.
  • Smart VoIP call Dialing Book: VoIP call Book could provide any application VoIP call to any type destination (Domain name/IP address, PSTN or PBX) or hunting number setting.
  • AC termination Impedance : 600/900 OHM and complex impedance
  • Polarity Reversal Detection: Type I and Type II
  • passNAT: This feature allow gateway to operate behind any NAT/Firewall device. There is no need to change any configuration of NAT/Firewall like setting virtual server.
  • Smart-QoS Guaranteed: This bandwidth management feature provide good voice quality when user place VoIP call and access internet at the same time. The gateway will start reserve bandwidth for voice traffic automatically when VoIP call proceeds.
  • Voice channels status display: This function display each port status like as onhook, offhook,calling number callee’s number, talk duration, codec.
  • H.323 MAC authentication : Providing H.323 MAC authentication to register H.323 Gatekeeper which need Mac address for authentication. (Note : Soundwin’s Embedded H.323 Gatekeeper provides IP address, H.323 ID, MAC address authentication policy)
contact: sam@soundwin.com
Image


Created by samchen0809, Last modification by samchen0809 on Mon 17 of Mar, 2008 [11:17 UTC]

Comments Filter

Decent FXO and FXS to SIP interface

by Chip Schweiss on Sunday 28 of May, 2006 [03:04:33 UTC]
I recently purchased from Soundwin their 402 model for testing. That's their 2 FXO / 2 FXS gateway. I've been testing it with Asterisk.

After facing horible echo problems with Sipura SPA-3000 and Digium cards on some networks, echo was a big concern on any FXO interface. The Soundwin has a 64ms echo cancellor (EC) that does a great job at EC. I've been testing it on a couple of POTS lines that no amount of tweaking on the Digium card could clean up. Thus far not even the slightest echo has been heard.

As an FXO port it has a couple of rough edges. Auto answer cannot be shut off, it can only be delayed up to 8 seconds. When it answers the call it plays an "answer tone" and then connects to the registered SIP server. This tone can be shut off, but it then plays a very short dial tone in its place. There is currently no way to make it simply ring the SIP server and wait for an answer before picking up the line.

When using g729 the comfort noise generator seems way too loud. Every time you talk you hear excessive static from the EC. Under ulaw it seems fine.

So far this is the best bang for the buck FXO port I've used.

They have everything from a single FXS & FXO gateway all the way up to 24 port versions. Each size is available 100% FXO, 100% FXS and 50/50 FXO/FXS.

UPDATE: (5/27/06) After trying it on a production system for a few days some significant problems have surfaced.

1. The echo cancellor has a significant problem with double talk. Instead of diverging it seems to over mute one or both sides.
2. After running for a while it will fail to authenticate with Asterisk when answering a call and play a busy signal to the caller. Switching to an unauthenticated stanza in sip.conf is the only thing that seems to keep calls coming in.
3. Periodically calls seem to be spontaniously dropped.

I have pulled it from the production system and all lines are connected to the highly reliable and predictable Sangoma A200d.

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