search for: voipstunt

voipstunt: "Forbidden - wrong password on authentication for INVITE to ...." I have paid, the password was not changed, ... I have no idea why. Is there anything what I can do to get this "failed" call over to another provider, so that the user can complete the call? (Dialstatus was an...

Hello everyone! I am using wine 0.9.30 with openSUSE 10.2 I've tried to install and run VoipStunt, and program installs with no error, but fails to start with the following output: dodo@Locutus:~> wine "C:\Program Files\VoipStunt.com\VoipStunt \VoipStunt.exe" preloader: Warning: failed to reserve range 00000000-60000000 err:module:import_dll Library gdiplus.dll (which is needed by...

The following is my dialplan for outgoing international call. What I want are: - when people dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, use voipstunt to dial out - otherwise, use my pstn to dial out. What I've found is when i dial 9011604xxxxxxx , 9011605xxxxxxx, 90114411xxxxxxx, 90114421xxxxxxx, it always use the pstn to dial out. Anything wrong with my dial plan? Thanks!! [outbound-oversea] exten => _9011604.,1,Macro(outgoingcall3,...

Hi, does any know why? i can make call out with my asterisk and voipstunt but i can't get call in on my voip in number i get rejected. if i use Sipura without asterisk i get in calls here is my sip.conf ---------------------------------------------- [general] useragent=nedi port=5060 context=default ;tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm...

...n, but remote party is disconnected. Basically, I am still on the phone, while remote party was disconnected. When I hang up, I get something like that: Apr 20 02:32:43 WARNING[4853]: chan_sip.c:8520 handle_response: Got authentication request (401) on unknown BYE to '<sip:0061385013322@sip.voipstunt.com>;tag=c9ebef50c90078c2c93eddc243d7352d6e04' Secondly, they charged me for calls to UK that was supposed to be free. And their customer service does not respond at all. Do they have a phone number I can call? Thanks, Rudolf

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking for a way, that covers all above. Voipstunt would get multiple accounts, whereby some are ONLY for free calls, and the others for paid...

...found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i call via a voip provider, bill seconds are counted incorrectly. Example: this call went to a pstn number New call from 551 ---> 94361abcdefg (context: internal) Dialed : SIP/00361abcdefg@voipstunt Call start: 2007-04-14 20:10:55 Answered : 2007-04-14 20:10:55 Call end : 2007-04-14 20:11:10 Duration : 15 sec Bill : 15 sec this call went to my ata from the sip softphone: New call from 551 ---> 505 (context: internal) Dialed : SIP/505|45 Call start: 2007-04-15 07:58:11 Answered...

...cho cancel capabilities) Currently the traveling users have to VPN in first which I am sure is adding extra overhead to the calls. I have yet to open my server to the Internet to be accessible to travelers without VPN first. I have done some testing with VOIP provider though my firewall to FWD and VOIPSTUNT. Where might SER help? Why are people using it with Asterisk? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. --- - --- - - - - - - - -- - - - --- - ------ - - --- - - -- - - - -- - - -

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codec...

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi I seem to have installed the G729 codec properly but can't seem to get it to work .. in the Asterisk startup I see .. [codec_g729b.so] => (Annex B (floating point) G.729/PCM16 Codec Translator) == Detected 1 licensed G.729 transcoders WARNING[1074408160]: File translate.c, Line 218 (calc_cost): Translator 'g729tolinb' does not produce sample frames. == Registered translator

[Jan 23 19:56:44] -- Got SIP response 300 "Multiple choice" back from 194.120.0.201 [Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to 'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:5060@default' (thanks to SIP/voipstunt-5c8c) [Jan 23 19:56:44] NOTICE[3439]: chan_local.c:483 local_alloc: No such extension/context 194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:5060@default creating local channel [Jan 23 19:56:44] NOTICE[3439]: app_dial.c:473 wait_for_answer: Unable to create local channel for call...

Simple as that please email me direct. voipviews@gmail.com Also looking for a U.S. DID provider as well as orig provider.

...e limits for temp buffer allocation) 13432 ZDaemon: freezes when exiting launcher 13459 Oberon game launcher (incl. buttons) does not work 13719 E-texteditor installs but freezes on run 14693 Program halts when on another virtual desktop 14718 Rappelz will not update itself 14873 Voipstunt crashes trying to connect to server 15417 Easy68k: Exiting without closing help window causes page fault. 16127 Mouse Buttons do not work in BBC Basic For Windows 5.80a 17035 RPG Maker XP 1.x: Script editor selects complete script when program gets focus 18195 Medal of honor pacific ass...