login | register
Sat 17 of May, 2008 [04:07 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Last Changes
Shoutbox
  • 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.
Server Stats
  • Execution time: 0.38s
  • Memory usage: 2.23MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.65

StanaPhone

http://www.stanaphone.com/

Free VOIP to/from PSTN service. Assigns numbers within various New York area codes.

Official client is a modified SJPhone soft phone.
Other SIP clients known to work include Cisco, Sipura, Grandstream, Atcom, X-Lite, SJPhone, etc.

All calls between StanaPhone users are free.
100 minutes per month free to 11 countries. Additional minutes can be purchased with a StanaCard.

Active user forum.

Sample Asterisk config for StanaPhone

VOIP Service Providers

Created by jht2, Last modification by papafox on Wed 04 of Aug, 2004 [01:49 UTC]

Comments

Comments Filter
Edit

* Fun

by Anonymous on Monday 31 of January, 2005 [00:34:24 UTC]
Sure seems they improved service a lot.
Edit

unable to receive calls/put credit on account

by Anonymous on Thursday 20 of January, 2005 [17:53:35 UTC]
They never ever billed my CC so i could make calls. (:sad:) Also they advertise free DIDs and i was never able to receive a call from that number.

Sounded like the perfect sip provider for me, but if it would only work... I've closed my account there and personally wouldnt recommend to other users looking for sip providers...
Edit

Can you register aserisk with stanaphone?

by Anonymous on Thursday 30 of December, 2004 [04:45:05 UTC]
I have not been able to register my asterisk box with stanaphone. I tried every single configuration sample i found to do this, but none work.
My * box is behind a NAT so I put what I believe is required for that situation yet it still wont register (i get a timed out error).
Anybody know why this can be happening? I just want to be able to receive calls for now...not interested in outgoing calls yet.
thanks,
Edit

Free 100 minutes per month discontinued

by Anonymous on Tuesday 24 of August, 2004 [23:50:49 UTC]
Announced by a Stanaphone representative in their online forum:

'The $2 credit for new accounts and renewal for existing accounts have been suspended until further notice.'
Edit

Multiple accounts?

by Anonymous on Tuesday 27 of July, 2004 [11:36:37 UTC]
Anybody know how to set up multiple accounts, I was hoping to "host" my brothers stana account as well as my own so that he can dial out to PSTN from our home system without using my minutes
Edit

Re: So So luck here

by Anonymous on Tuesday 27 of July, 2004 [11:34:11 UTC]
Change dtmfmode=rfc2833

Edit

Re: So So luck here

by Anonymous on Thursday 15 of July, 2004 [18:04:37 UTC]
I get kind of the same thing. WARNING1242848048: dsp.c:1467 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833

Any fix for this that someone has found?

So So luck here

by wsuff on Friday 09 of July, 2004 [23:28:05 UTC]
Seems to call out and dial in properly but I can't seem to get DTMF to work.
dsp.c:1467 ast_dsp_process: Unable to process inband DTMF on 2 frames
Since they require inband

Yes

by haiqu on Saturday 26 of June, 2004 [05:30:07 UTC]
It works here. Have made calls to Germany, others making calls to UK, Canada, etc.
Edit

Re: call to non-us countries

by Anonymous on Thursday 24 of June, 2004 [19:02:32 UTC]
Yup, called HK.

It works, but the quality is not good ....(:frown:)

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2008 VOIP-Info.org LLC

Powered by bitweaver