similar to: WebPhone

I am looking for a way of not to install a softphone, preferable as a link on a web site to a webphone on MY SITE !!! Has anybody an idea for that? AJAX? bye Ronald Wiplinger

Greetings everyone, I've been trying to make this java based webphone work for everybody visiting my website, but seems like for many users it doesn't work. In order to get a better idea what is the success rate of this webphone, I would appreciate help from anybody who could make a few calls from it within North America and if it doesn't work, send me what error you get, or if it

Hi, someone can help me, I am ot able to find the way how to user Ajax.updaterto test if the request give some positive or negative result. I am able only to return the result inside a div. An example is appreciated. _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org

I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if I call the traffic still go throw the asterisk. How come? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm

Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I installed this program but I am not able to configure, it does not want to work. Someone can help me?

Hi, We are using MozIAX as a softphone with a USB headset and are making outbound calls using IAX with ulaw encoding to our voip provider. We're running asterisk 1.4 Users are complaining that the ring tone generated by asterisk is much louder than the voice call once connected. They are having to turn the volume down to avoid being deafened by the ring tone, but then have an unacceptably

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm

There is a way to force the order of the codecs in the sip.conf since the allow seams to let know only the accepted codec. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/619b8f2b/attachment.htm

Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

Is anyone doing this or has anyone tried? The thin clients are running WindowsCE, a browser, and 300mhz. They are Wyse units. I wonder if anyone has any practical advise or can recommend the best phone or method to load a stable softphone on one of these boxes? Thanks, Steve Totaro

I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? Regards AK

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox 1.6? jslib and moziax install through Firefox correctly - at least that is the message I get. I am able to log into the IAX Phone on Windows, however I get an error stating: -------------------------------------------------- FATAL ERROR: no connection to "network_client". MozPhone will stop now!

Hi, I saw one site (bubbleshare) that it is able to caputer the click on the log in link, however, I cannot understand how they can do that Someone can explaint it to me? Thank you _______________________________________________ Rails-spinoffs mailing list Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org http://lists.rubyonrails.org/mailman/listinfo/rails-spinoffs

Hi, I'm using a script in order to send out my faxes with the application txfax, therefore, I do not know how to see if the faxes are sent. Any idea?

I have this simple context I am register to an external provider and when I am not home I would like to transfer the phone outside The problem that the call goes in loop I cannot understand why. Can you figure out my error? Thank you sip.conf register => user:pass@provider/400 [inside] exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT) exten => _4X.,2,hangup -- Executing

Hi Neofita, Doug and All. I think I've the same problem but I don't know if it's related to the bug suggested below. I try to explain my behavior: - I dial the voicemail extension. - I hear: "You have 1 new message. Press 1 for new messages, press 2 for... or # to exit" (I listen the complete message or most part of it) - I press 1 - I can hear the first recorded message.

Hi, I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications. You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example) ? /**************************/ simple HTML code example: /*************************/ <html> <head>