similar to: [Asterisk-Dev] sip messaging (tested on eyeBeam) support

Displaying 20 results from an estimated 10000 matches similar to: "[Asterisk-Dev] sip messaging (tested on eyeBeam) support"

2005 Jun 13
1
presence and video conference
Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there
2005 Feb 01
1
FW: Messaging with * and eyeBeam
-----Original Message----- From: Ferguson, Michael Sent: Tuesday, February 01, 2005 11:35 AM To: 'asterisk-users@lists.digium.com' Subject: Messaging with * and eyeBeam G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I called xTEN but they preferr to talk to telcos and ISP's
2005 Feb 01
0
Messaging with * and eyeBeam
G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I called xTEN but they preferr to talk to telcos and ISP's purchasing hundreds of the eyebeam software... Kind-a stuck here. I already have * happily running and taking care of business on my Windows network. I also have Windows IM running on
2006 May 03
6
ruby on rails international & BIRT integration?
Hello, I see, that Rails is quite english-centric. I am developing some webs, that are not primarily in English. I have a few questions: - besides turning of plurals, what should I take care? How to use utf-8 for all data and converting it from local charsets to utf-8? - how do I make my page multilingual (i.e. adding english support later)? Is there something like gettext support? Is
2005 Jul 04
1
[Asterisk-Dev] presence and IM again, want to develop a working "hack"
Hello, I was again asked to try to add support for presence (SUBSCRIBE/NOTIFY) and IM using SIMPLE. I have few questions: a.) are there any, at least partial projects, patches, anything, that at least partly implements presence and/or IM to chan_sip? I don't care about presence on other channels, I have one SIP client per user. I've read this list's archive several times and
2005 Aug 20
3
[Asterisk-Dev] IM patch
Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
2005 Sep 02
1
Asterisk and Eyebeam
What's the status on using eyebeam with Asterisk, does it still require a patch to Asterisk to support the video component? I'm intererested in starting to use Video and audio telephony but wary of anything that requires patches. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM:
2005 Jan 31
0
Eyebeam Vs. Windows Messanger,
G'Day All, Eyebeam has gotten my interest but I do not have a "high-altitude" view of its interraction with *, therefore my questions. I already have * happily running and taking care of business on my Windows network. I also have Windows IM running on my exchange server and users a IM'ing quite a bit - all internal though-. While I would prefer to use * and eyebeam, I am quite
2005 Oct 13
0
polycom soundpoint ip600 problem
Hello, I have a polycom ip600 and eyebeam. When I call from polycom to eyeBeam, everything, including audio works. When I call the other side (from eyeBeam to polycom), I get no audio. In both cases, eyeBeam shows the same codec: g711u. Also sip show channels shows ulaw codec for both sides and correct addresses. I have canreinvite=no. I don't know if it's important, but asterisk
2005 Sep 07
1
presence settings and Eyebeam
What is the proper way of adding hints to multiple extensions? In my case extensions are the same as the sip usernames, while as per http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence exten => 1234,hint,SIP/1234 works, exten => _1XXXX,hint,SIP/${EXTEN} doesn't. Not sure if I can even use ${EXTEN} here... Any hints? Vahan -------------- next part -------------- A
2008 Oct 22
3
asterisk video
hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start
2005 Jan 30
0
xten x-lite eyebeam
In an attempt to eliminate audio echo I upgraded one side of a working x-lite to x-lite connection to eyebeam. No joy, and what was worse is the audio was even worse now - just noise. Ok, I upgraded the other side to eyebeam and same thing. I'm not even using video (will enable it in sip.conf later, one change at a time). The connection looks something like: eyebeam client
2007 Dec 04
4
enable eyeBeam to accept only one call
Hello I'm using eyeBeam, and Asterisk keeps sending my clients a second call, when they are still in one call (because eyeBeam has lots of channels). I was using X-Lite (with 3 channels) and Asterisk never sent the client a second call. How can I force Asterisk (or eyeBeam) just to send one call at each time. Is this a configuration I need to do in eyeBeam or Asterisk? Thanks Regards Joao
2008 Sep 28
1
G.722 between Eyebeam and a Polycom IP650
Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the
2009 Jan 29
2
Eyebeam or Xlite
Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090129/4011be6c/attachment.htm
2006 Apr 18
2
eyeBeam + ASterisk 1.2.7.1 + Instant Message
Hi, I'm trying to find how to configure Asterisk 1.2.7.1 to allow two EyeBeam (3015c) to send Instant Messages between them... But I cannot find anything that explains how to do it! Anybody as a clue? is it possible? Now, when we try to send an Instant Message in the eyeBeam it says: "User not available". In asterisk console appears a message saying: ------ Apr 18 17:13:22
2005 Jul 12
1
Little doubt on Asterisk and EyeBeam
Hi, Recently I been using EyeBeam with Asterisk. I can make calls with video, I'm using two PCs with EyeBeam but I noted that I can't enable only one client to stream video, I mean if I start streaming video on one client the other one doesn't receive any video until it starts streaming video itself. So I cannot have unidirectional video calls. I remember this problem with
2005 Aug 16
1
problems with eyebeam - video phone
I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;
2010 Jun 10
0
Eyebeam hangs when you dial an unavailable number
I am having problems with Eyebeam when the user dials a number that is not available. This problem exists with both Asterisk 1.4 and 1.6 using Eyebeam or Xlite. The problem seems to be that when the soft phone receives the 503 Unavailable response it will not be able to dial another number for a few minutes. Anything you dial will say that the number is unavailable and it will not even send the
2005 Jul 06
1
g.729 codec -- open source?
Hello, is there an open-source implementation of G.729 codec for use outside of US? I know it's a patented codec, but since there are usually no software patents outside of the US, I don't care about the patent license. I could use open-source implementation of the codec, if there was some. Any ideas? Sincerely, Juraj Bednar.