similar to: Re: [Asterisk-Dev] MS Live Communications 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

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.

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:

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

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

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

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 ***

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.

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

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

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

Muchísimas gracias Carlos, de verdad que te agradezco la ayuda, pero no es lo que voy buscando. Quiero colocar en el eje de abscisas la secuencia temporal de los meses, es decir, agosto septiembre, octubre, etc? pero no las fechas de las toma de datos, sino que aparezca la marca de un mes, y la siguiente marca sea la del siguiente mes, etc?, y además que las muestras estén separadas de acuerdo con

Muchas gracias Carlos, previo a mi correo, entre las pruebas que hice estaba una parecida a la que apuntas de la siguiente manera: attach (Libro1) plot (xbar~as.Date(fechas,"%d/%m/%y"), ylim=c(400,660), xaxt="n", type="b", pch=19,cex=1) xlabels<-strptime(fecha,format="%d/%m/%Y") axis.Date (1,at=xlabels,format="%b-%y")

Hi list, I experiencing a strange behaviour when transferring a call. The use case is like this: - Incoming call from Zap/1-1 - Routed to SIP phone SIP/1001 - The called user (SIP/1001) wants to redirect the call and presses "#" - IVR (default setup) says "Transfer" and user gets dial tone - User dials 1002 - IVR says "No such extension - please try again" ??? It

I tried to compile pam_ldap-184 but it gave lots of error msgs. BTW, I have successfully compiled nss_ldap-255. For pam_ldap, my configure looks like: ./configure --with-ldap-lib=openldap --with-ldap-dir=/usr/local --with-ldap-conf-file=/usr/local/etc/openldap/ldap.conf and the following is the configure output --- start of configure output --- creating cache ./config.cache checking host system

Hi, I have been experimenting with NAT and Asterisk a bit now. Though I have made progress along the way, I have come across the following problem. I'll really appreciate if anyone can provide me any help or pointers. Thanks! Successful Scenario: ------------------- All sorts of NAT calls are successful with full two-way media when both end points are locally subscribed users. Problem

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

Hey all, on the distcc mailing list, a thread about load balancing got a bit out of hand, and we started thinking about moving fsh-like connection caching into ssh itself to get rid of the overhead of starting up the python interpreter to run rsh. (Interestingly, mit's "rex", described at http://www.lcs.mit.edu/publications/pubs/pdf/MIT-LCS-TR-884.pdf, considers connection caching

Hi, I am in a predicament and any help/pointers would be appreciated. We are using chanspy to listen in on conversations. We are doing this via a web interface. The web interface lists all the ongoing calls. We click on a call and then my local phone rings allowing me to spy on the session I clicked on. But "most" of the time, when I start listening in, the two parties that are in

As previously noted, I need to replace a UPS and want to do my product research in public so others can benefit. The information I elicit on this thread will be merged into the UPS-HOWTO. I went to my local computer-equiuipment big-box store, a place called MicroCenter in Ardmore, Pennsylvania. I looked at the range of "UPS" devices they sell and was *completely* confused. Bear in