Displaying 20 results from an estimated 500 matches similar to: "Live Communication Server and Asterisk"
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
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 Jun 29
1
Asterisk/SER/Call Manager
Hi all,
I have Asterisk talking to my call manager 4.0 with SIP trunk as mentioned
in the wiki. I also have SER talking to Asterisk. I need the SER talking
to my Call manager. The reason why CCM cannot talk to SER is because SER is
a on a public ip address, and CCM is on a private ip address.
The asterisk how ever has 2 nics, which talks to both and external. Is it
possible to allow
2005 Jun 01
2
IAX2 analog telephone adapter
Hello All,
I am looking for a IAX2 analog telephone adapter, just want to ask your
views on which ones are bad, good and the best.
Thanks in advance,
Dinesh Birlasekaran
Network Engineer,
ComIT, Institute of Molecular and Cell Biology
61 Biopolis Drive, Singapore 138673
HP : 92962676 DID : 65869804 Fax : 67791117 Email :
dinesh@imcb.a-star.edu.sg
WWW: www.imcb.a-star.edu.sg
2004 Nov 30
3
Cisco Asterisk Integration
Hello All,
I have managed to get my cisco and asterisk able to talk to one another I
think. But cannot make a call from a phone behind call manager to the
asterisk server.
I have followed the cisco asterisk integration on the wiki.
I have also setup a number 3000 for dialing for current local time and date
on asterisk. I can call from a sip phone behind asterisk, no problems. The
problem
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
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
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
***
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
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
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:
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 Jun 02
0
Connecting Asterisk with Microsoft LCS (Live Communication Server)
Hello,
im trying to connect LCS to asterisk which will act as pstn gateway for LCS.
Microsoft system supports only SIP TCP connections but asterisk UDP.
im was searching about conversion beetwen TCP and UDP and i found that SER
can do that but i don't know SER and my trying to configure SER fails.
is there any other possibility to connect this together?
Maybe someone has correct
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
2001 Oct 31
0
implementation of subnet bandwidth manager for linux
Hi,
Is there an implementation of subnet bandwidth manager that is available
for linux? I know that there is one available in win2k and also, that
internet2 QBone is working on bandwidth broker but I haven''t been able to
locate any implementation of subnet bandwidth manager (SBM).
Thanks,
Deepak
--
=====================================================================
Deepak Bansal
2012 Feb 04
8
Potential memory leak in sshd [detected by melton]
Hi all,
After the memory leaks (bug 1967
<https://bugzilla.mindrot.org/show_bug.cgi?id=1967>) I reported in
bugzilla are fixed,
I also applied melton(http://lcs.ios.ac.cn/~xuzb/melton.html)
to detect the potential bugs in sshd (openssh-5.9p1).
The url below is the index of bug reports that are checked as real
bugs manually.
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
2011 Dec 30
7
[Bug 1967] New: Potential memory leak
https://bugzilla.mindrot.org/show_bug.cgi?id=1967
Bug #: 1967
Summary: Potential memory leak
Classification: Unclassified
Product: Portable OpenSSH
Version: 5.9p1
Platform: All
OS/Version: All
Status: NEW
Severity: normal
Priority: P2
Component: ssh
AssignedTo: unassigned-bugs at
2007 Jun 06
1
Polycoms lose registration and won't re-register
For the last few months we have intermittently been experiencing some very
strange registration problems with certain polycom phones.
Here is some background information:
I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8
servers at different locations. Each phone is on the same network (and
subnet) as the server it connects to. There is no NAT or anything else
strange
2009 Jun 13
2
Polycom registration errors
I'm evaluating using Polycom phones for our call center and I've set
up my first phone (a SoundPoint 560) to give it a try.
The phone is working and can successfully place and receive calls.
But every minute, there's an error in the log file:
chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for
'192.168.200.99' - Username/auth name