Displaying 20 results from an estimated 20000 matches similar to: "Connecting Asterisk with Microsoft LCS (Live Communication Server)"
2005 Jun 22
1
Fwd:protocol TCP/UDP question
can you help me to configure lcs2005 with asterisk...
I use SER to resolve the problem that there is for communication protocol...LCS uses tcp, Asterik UDP.
Someone, knows how to do the configuration beetwen LCS and SER , SER and Asterisk? the function of asterisk is SIP-PSTN Gateway for the LCS PC-phone communication??
or there is a way to configure asterisk to accept tcp communication from
2006 Feb 19
0
Live Communication Server and Asterisk
Has anyone have interfaced this successfully? I came to know from M$ that
Genesys GETS can be used to interface asterisk. I have interfaced Cisco
call manager to asterisk/ser but for my final setup I would like to have a
LCS talking to a CCM, without having the Genesys GETS is I don't have to.
Has anyone been playing around with this? If so I would really like to hear
some advise.
2006 Sep 11
0
[Serusers] MS LCS 2005 / SER / Asterisk Integration
Hi to all,
I read
http://www.voip-info.org/wiki/view/MS+LCS+2005+%252F+SER+%252F+Asterisk+Integration
Is it possible to use ser as a presence server instead
of LCS 2005 ?
Harry
___________________________________________________________________________
D?couvrez un nouveau moyen de poser toutes vos questions quelque soit le sujet !
Yahoo! Questions/R?ponses pour partager vos
2005 Aug 11
1
MS Live Communication Server
Hi List!
does anyone played around with the LCS and Asterisk? Because the LCS is
doing no RFC compliant SIP, i wonder if it can work. Google couldn't
tell me. If someon heared about that, please let me know.
The fact i figured out is that the Border Controler from Jasomi can be
used as a gateway from MS-LCS-SIP to regular SIP. But that is not really
handy and expensive too.
Thank you
2005 Dec 05
3
PRI indications.
Hello,
i have succesfullu setup asterisk with Sangoma E1 card, evrything works well
but i don't know how to pass indications from telco switch to the user - when
users call bad number telco switch shuld talk "unallocated number" but its only
send PRI_CAUSE 1. How to pass voice indications thru asterisk to clients?
My /etc/zaptel.conf:
span=1,0,0,CCS,HDB3,CRC4
dchan=16
2004 Jul 04
0
LCS multiparty conferencing commercial opportunity
Hi this is just a heads up about an opportunity for commercial Asterisk
experts. I don't know if this even possible but don't see why not and it
is way beyond my capabilities so thought I would pass it out to the
list.
I've been looking into Microsoft Live Communications Server over the
past few months for one of my clients, it's the same as ms messenger but
for closed user
2005 Aug 15
1
Re: [Asterisk-Dev] MS Live Communications Server
Search google with "sip pstn site:www.microsoft.com"
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to
next hop: pstngw ip address
Step2: patch your asterisk chan_sip.c to support TCP
Step3: configure
2008 May 05
4
microsoft office communicator 2005
Hi! im trying tu run "microsoft office communicator 2005" and i cant
resolve this:
fixme:ntdll:NtConnectPort (0x1434f8,L"\\RPC
Control\\epmapper",0x33ecd0,(nil),(nil),(nil),0x33ecf8,0x33ece0),stub!
i google it all nigh long and i just cant find the way!!!.
I need to connect to LCS 2005 because my company switch from Jabber to LCS.
I tried pidgin and miranda-im+sip but didnt
2004 May 18
0
.(±¤.°í) Áú¿°,³Ã´ëÇÏ,»ý¸®Åë µî ¿©¼º°í¹ÎÀÌ ÀÖÀ¸¼¼¿ä? LCS°¡ ÀÖÀݾƿä. ÀáÀÚ±âÀü¿¡ Áú¾ÈÂÊ¿¡ ³Ö°í ÀÚ¸é ºÒ¼ø¹°ÀÌ »ý¸®Ã³·³ ³ª¿À°í ¿©¼º°í¹ÎÀÌ »ç¶óÁý´Ï´Ù[[RANDOM]]
<head>
<meta http-equiv="content-type" content="text/html; charset=euc-kr">
<title> L.C.S</title>
</head>
<body>
<table align="center" border="1" width="468" bgcolor="#F9DEDE" bordercolorlight="red">
<tr>
<td width="462" height="211"
2007 Jun 06
5
TCP<->UDP SIP proxy?
Hello,
One of our faculties have Microsoft's LCS and would like to connect it to
our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS
talks SIP over TCP with TLS. Anyone can recommend a gateway between these two
protocols?
Thanks! __Yehavi:
2009 Jun 17
2
Difference beetwen element in the same column
Hi, i have this file
pressure,k,eps,zeta,f,velocity:0,velocity:1,velocity:2,vtkValidPointMask,Point
Coordinates:0,Point Coordinates:1,Point Coordinates:2,vtkOriginalIndices
0.150545,0.000575811,0.0231277,0.000339049,-0.0193008,0.00318629,-6.24066e-07,5.39599e-05,^A,7,0,0,0
0.150546,0.000782719,0.0226157,0.000497957,-0.0192084,0.00367781,5.09813e-06,5.90689e-05,^A,7,0.0003035,0.000225,1
2005 Jan 24
2
PSTN and Asterisk
Hi quys,
I look for a solution for interconnection beetwen PSTN and VoIP.
My application have to treat few protocols comming from PSTN lines and mixing data , dtmf and voice.
Can I use Asterisk for :
PSTN ----------> Asterisk (converting analog call to IP) ----------> MyApplication ( translation protocols and do some works with incomming data)
What hardware I can use for this?
Do use
2008 Mar 10
11
Microsoft Office Communications Server
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Has anyone done any integration with this?
All I know so far is that it appears to use some non standard form of SIP.
Any pointers?
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News -
2003 Dec 22
2
Sipura 2000 configuration.
Ok here is another problem I have run into.
I have a Sipura 2000 and I have been able to configure line 1 with only
one small problem. But I can't get the line 2 working with asterisk.
Here are samples of my sip.conf and extensions.conf. If I disable line
1 I can then get line 2 working. Is there a sample configuration for
the Sipura to get both ports working with Asterisk.
Sip.conf
2006 Jun 01
1
connecting asterisk to pstn help
Hello Masters
Here i going explain what Iam doing and where i need help ..
Iam running Sip Express Router ,Asterisk, on same box (for
testing) my Sip express router is working fine and i can accept global
register requests with valid account and in front of Sip express router
(SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams
between nated clients
2011 May 08
3
Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hello all,
I have installed the .deb packages of the Asterisk v1.8.3.3 from the
upstream project on my Debian GNU/Linux Squeeze server and bought the
Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS
exercise. After setting up everything and trying to fix this problem,
I am still getting a 401 Unauthorized SIP message. So as of this
writing, I still cannot successfully REGISTER
2016 Sep 01
4
Microsoft R Open 3.3.1 problema
Estimados
Microsoft R Open 3.3.1 me está dando problemas, por ejemplo rbind.
Los resultados son extraños, por ejemplo muchas columnas cuándo debería ser una sola sonde tomo solamente la primer columna de varios data.frames, como un arreglo de n x n donde los n son números ?grandes?, cuándo debería ser solo 1 x n .
¿Alguno tiene problemas? Está imposible para trabajar.
Javier Rubén Marcuzzi
2009 Apr 24
2
Thanks for the lenny-cran AMD64 ports
Hi.
I would like to thank Johannes, Piet and others for the lenny-cran AMD64
ports.
I have a question about r-recommended from lenny-cran which I just
installed:
hotelling:~$ apt-cache policy r-recommended
r-recommended:
Installed: 2.9.0-1~lennycran.0
Candidate: 2.9.0-1~lennycran.0
Version table:
2.9.0-1 0
600 http://debian.lcs.mit.edu unstable/main Packages
***
2000 Apr 04
0
Obscure bug....?
Dear all,
I've been struggling for days now with a piece of code that I have posted
here before, that has a really obscure bug. I think I may have isolated
it, but I have no idea what it is.... It might also be a bug in R I
guess, as it seems that one or several of list elements are not passed
when a function is called, but quite rarely.
I have been hacking rather wildly on the histogram
2004 Dec 11
2
ACK from asterisk not matched to transaction by SER / LCS2005
For reasons unknown to me, SER and subsequently a Microsoft Live
Communcations Server 2005 seems to have problems, matching a SIP ACK
request from asterisk to the ongoing SIP transaction, I have attached
the complete log, but the essential lines are:
13(2894) DEBUG: RFC3261 transaction matching failed
13(2894) DEBUG: t_lookup_request: no transaction found
13(2894) SER: forwarding ACK