similar to: i sided recordings in asterisk 16.10

Hello Can I use Cisco 3950 on Asterisk ? Thanks.

Hi, I have latest Elastix 64 bit setup and running fine (Asterisk 1.6.2.13) I would like to customize the file name of call recordings: /var/spool/asterisk/monitor/20110511-110858-1305126538.912.wav I would like to include the extension number in the file name. Did a lot of googling but not much help. Pls advice. Thx Sans -------------- next part -------------- An HTML attachment was

Hi, I''ve been looking for a light weight alternative to rails for a few small projects, and just came across merb, which looks perfect. The only issue is that merb seems to be tied to mongrel, and I have to deploy to our internal infrastructure which uses FastCGI. How difficult would it be for me to modify merb to support a fcgi interface (actually a rack interface - rack is

Eli Gottlieb <eligottlieb at gmail.com> writes: > I compiled and installed it to the prefix /usr, but that's not the > issue. Once I actually compile and install LLVM with CMake by hand, I > get the share/llvm/cmake stuff installed correctly (can those files be > included in "normal" builds, or will LLVM switch to CMake as its > primary build system?). Now

Zitat von Israel Gottlieb <isrlgb at gmail.com>: > If you the c option in the dial command it will send answered > else where sip message to the phone and most ip phones understand that > The cell will always display a missed call? I'm very sorry, but I can't understand what you mean... Could you explain, maybe with an example? Thanks Luca Bertoncello (lucabert at

I have a directory dumps on my laptop containing several dumps of various levels. local-0-2007-03-03.gz local-4-2007-02-12.gz local-4-2007-02-19.gz local-4-2007-02-26.gz local-4-2007-03-05.gz local-5-2007-03-04.gz local-5-2007-03-06.gz local-5-2007-03-07.gz local-5-2007-03-08.gz Naturally the level-0 is the largest and rarely changes. On the target (access.cims.nyu.edu) I have a directory

After I actually get everything compiling, install the library, and load it from my Java program, I get the following: > Exception in thread "main" java.lang.UnsatisfiedLinkError: > /usr/lib/libjllvm.so: /usr/lib/libjllvm.so: undefined symbol: > _ZTVN10__cxxabiv120__si_class_type_infoE If I have to guess, this means that the CMake stuff given is linking to the C++ libraries

I am new to using R and would appreciate some advice on which books to start with to get up to speed on using R. My Background: 1-C# programmer. 2-Programmed directly using IMSL (Now Visual Numerics). 3- Used in past SPSS and Statistica. I put together a list but would like to pick the "best of" and avoid redundancy. Any suggestions on these books would be helpful (i.e. too much

Hi Folks, I''ve been trying to get camping up and running on dreamhost for the past six hours, but have had no success. I''d really appreciate any suggestions! The camping gem is on the dreamhost server, but just to make sure it wasn''t a dependency issue causing all the problems, I installed camping + the full set of dependencies in my own gem directory (both are on the

Eli Gottlieb <eligottlieb at gmail.com> writes: > So you're saying that the default CMake build of LLVM creates static > libraries that got linked into my shared-object and now require me to > link in everything they require myself? Shouldn't the linker be able > to track down C++ runtime for this? You told CMake to manage your shared library as if it were a pure C

On 2003-04-26 at 00:42, Jim Gottlieb (that's me) wrote: > [ccmenu] > exten=s,1,Ringing > exten=s,2,Wait,2 > exten=s,3,BackGround(5045) > exten=s,4,Goto,outtrunk|17005554223|1 ; if they just wait > exten=_X,1,Goto,outtrunk|17005554223|1 ; if they press 0-9 > exten=_*,1,Goto,outtrunk|17005554223|1 ; if they press * > exten=_#,1,Goto,outtrunk|17005554223|1 ; if they

Zitat von Israel Gottlieb <isrlgb at gmail.com>: Shalom, Israel! > Using chan_sip you need to create another ?user aand then dial both > > Using pjsip you can connect 2 devices Thank you. Unfortunately it seems that I don't have pjsip available as package on the OpenWRT where I installed Asterisk... :( I'll create another user. Thanks Luca Bertoncello (lucabert at

Israel Gottlieb <isrlgb at gmail.com> schrieb: > At the end of the Command you could use options one of them is the c (not > apital) which sends a cancel event to the phone > http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Shalom Israel, unfortunately it does not work as expected... I wrote: exten => _00493512222222,n,Dial(SIP/00493512222222&SIP/00493511111111,,Rc)

Any chance of say digits being extended to recognise "*" & "# " ?? Heck these are digits on a normal keypad :-) Gary .

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

We have bought and installed 10 channels worth of G.729. We reset our ATA186 boxes to use this encoding, and it seemed to be working. However, we analyzed the data stream and saw that the ATAs were sending about 4 kbps, while the stream from asterisk was still 64 kbps, which says to me that asterisk is still using ?-law. I looked in the sample sip.conf file and don't see any place to set

I just loaded from CVS this afternoon and in the debug output I see... DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: m on Zap/16-1 DEBUG[76820]: File chan_zap.c, Line 3254 (zt_read): DTMF digit: u on Zap/16-1 I knew about DTMF 0-9, A-D, *, and #, but I didn't know about m and u :-).

Hi. Can anyone point me to some docs detailing how to set up a connection with Level 3 Communications? A customer of ours wants us to terminate some inbound service via Level 3 to our asterisk server. I've tried all sorts of settings but nothing yet has worked. SIP debug shows a 407 Proxy Authentication Required error. I haven't been able to find anything on the web, and the techs at

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160810/7e14a4e0/attachment.html>

I would like to send the CDR records from all our machines around the world to a single database. But I need the hostname included with each record for monitoring purposes. Is there a better way than using the userfield and adding SetCDRUserfield for every call to set the userfield to the name of the host? Thanks...