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Wed 01 of Aug, 2007 [09:11 UTC]

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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
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Phone Numbers

You can call the following VoIP numbers for free. From a softphone, just copy&paste the full SIP address (sip:user@domain.com). From a hardware SIP adapter, copy&paste the full SIP address into your speed dial web page. All of these numbers worked in June 2007.

Notes:
  1. To dial from a hardware SIP adapter, you can usually paste the entire address into your phone's speed dial web page. SIP adapters cannot call any IAX numbers.
  2. G.711 is the usual codec that VoIP phones use, but the WiSiP phone works better with G.729-only selected. Calls from a WiSiP phone to VoIP numbers that don't support G.729 will ring and connect, but you won't hear anything after that.
  3. Suggest testing from pulver.communicator or StarTel.pt with the X-Lite softphone. Please list only *open* VoIP phone numbers and Free VoIP Networks, meaning that you have successfully made a call to that number from a SIP proxy located in a different domain. For example, you can call sip:xmission@pbx.xmission.com from X-Lite which registered in the StarTel.pt domain, so the xmission SIP number is open. On the other hand, Vonage numbers can only be called for free from other Vonage phones, so Vonage is NOT open to calls from other SIP proxies.
  4. If you install SJPhone, it will make SIP a "registered protocol" in Windows, so that clicking on one of the above 'sip:' links will automatically open SJPhone and start the call.
  5. http://www.trxtel.com has discontinued free calls to 1-800 numbers, "Due to recent legal activity against some phone companies and other operators on VoIP terminated calls"

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IP Phones

Created by human455, Last modification by Joao Pereira on Tue 12 of Jun, 2007 [13:39 UTC]

Comments

Comments Filter

Re: US DIDs with E911

by David Chappell on Wednesday 11 of October, 2006 [02:36:45 UTC]
Isn't this spam? Is there a way to report comment spam?

US DIDs with E911

by fonemantx on Wednesday 04 of January, 2006 [22:30:22 UTC]
We can provide US DIDs with E911 for over 99,000 US NPA-NXXs. We also have white route Mexican DIDs. www.i3net.us/carrier.html

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