Displaying 9 results from an estimated 9 matches for "webphone".
2009 Aug 30
1
Help me testing this webphone at www.VisionVoIP.com
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 e...
2006 Apr 26
6
I am looking for a webphone on MY SITE
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
2006 Jun 27
5
WebPhone
Hi,
someone know a good webphone, possibily a free one
Thx
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2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
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>
2014 Jul 02
1
Webrtc Not acceptable here
...uot; for peer
1061
== Using SIP RTP CoS mark 5
[Jul 2 19:57:04] WARNING[26672][C-00000071]: chan_sip.c:10648 process_sdp:
Can't provide secure audio requested in SDP offer
If any more information is needed please let me know
My goal is do do peer to peer calling with asterisk+webrtc (i.e. webphone)
--
Regards
Sameer Rathod
8109413462
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2007 Feb 18
3
chan_sip.c:1968 create_addr: No such host:
I have followed all the install note for A2billing and have everything installed and configured and my asterisk works except the callingcard application.
Added the following
[callingcard]
; CallingCard application
exten => 777,1,Answer
exten => 777,2,Wait,2
exten => 777,3,DeadAGI,a2billing.php
exten => 777,4,Wait,2
exten => 777,5,Hangup
I am using 777 as the calling card
2010 Jun 28
2
restricting sip users to a certain useragent
Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will
2008 Jul 07
8
US T1 Hangup Detection
We are in the process of preparing to move our Asterisk server to a
Digital T1 interface card instead of a analog card (via an Adtran
which is now connected to the T1). I did a preliminary test the other
day and hooked the T1 line up to the T1 card, bypassing the Adtran.
This worked rather well I must say. The two issues I ran into are:
1) Caller ID is not working even though I enabled
2009 Aug 31
0
asterisk-users Digest, Vol 61, Issue 85
...Re: GoToIfTime : how to define sep 25th till oct 10th ?
(David Backeberg)
5. Re: GoToIfTime : how to define sep 25th till oct 10th ?
(MeetMeCall)
6. unable to execute command from (DOCAS DUDU ZULU)
7. Re: unable to execute command from (Alex Balashov)
8. Help me testing this webphone at www.VisionVoIP.com
(Zeeshan Zakaria)
9. Re: Asterisk 1.6.0.14 and 1.6.1.5 Now Available - pbx/lua.c
changes (Tilghman Lesher)
10. Asterisk/app_rpt and bandwidth (Michael Maxwell)
11. Re: PRI worked fine for months, nowit stopps working after
2-3 hours (Loic Didelot)
12...