search for: twinklephone

...gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for ISDN also works good. But when I make a connection through IAX to another asterisk (having allow=g722 to really use G.722 in IAX) the voice is 'broken'. I also work on G.722 for twinklephone and encountered a special thing about G.722: It has a sample rate of 16000, but it announced as 8000 in SDP. Since I have similar problem with my G.722-twinkle implementation, it looks like the RTP and/or jitterbuffer code has a problem with that. Did I miss something here or is this really a bug...

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

...this request does. I think the mode somehow maps to the bitrate, but I cannot find a table that tells me which mode maps to which bitrate for nb, wb and uwb codecs. And what does it mean when mode=any? Can someone explain me what the mode indicates? Best regards, Michel -- Michel de Boer www.twinklephone.com

The receiver could send the packet list that the he lost directly via p2p to the clients. 1. it isnt much data to send. 2. you safe the way throught servers. Now the sender sends the data again to the server. Or if it isnt much to send he could send it directly.... How about that? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not sure if it's an issue but my voip provider uses port 80 for sip instead of 5060. In asterisk, I can make outgoing calls through my voip provider to pstn li...

Anyone have an rpm for twinkle? http://www.twinklephone.com/ A new sip softphone.