similar to: h323 guide for asterisk

Hi people! I am not getting smart getting asterisk 1.6 behind a NAT to run. 1. I enabled IP forwarding on debian linux 2. told asterisk in "general" that he is behind NAT and mentioned him his external static IP Adress as well his domain in the outside world. If a client who is connected with a DSL modem calls me, a grandstream module in the LAN behind the router, in the same network

Hi! I have problems building asterisk 1.6.0.6. ./configure --prefix=/usr make gets me: enerating embedded module rules ... [CC] extconf.c -> extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file

Hi people! I want to save the 1st letter from the ${EXTEN} variable. I don't want to trim it, I want to RESAVE it into a new variable. Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0 I would thank you for all advises. Tamer

Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer

Hi people! I am coding a special sollution for that I need to know if I can send AT commands in the extensions.conf, to one subscriber. Is there a way doing this through asterisk 1.6 ?! For sure anybody of you, would as why I want to do that. I want to speak to my endsystem directly with AT commands. For any advise, I would thank you kindly. Tamer

Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49,

Hi! I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: 1. The Base station of the DECT cordless phones, is connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want to place and receive as many calls at the same time through 1 SIP

Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html

Hi, R-users, I met a problem: Items:[Anna 'moi =) akku loppu joskus 4ltä. Kestää kauan nää..'\tAmer, Tuusula (0:20)\t20\t12\t16\t00\t00\t11]/Anne 'Ei jakoa,uus päivä muistio et 4n niin peruin. Hups'\t (0:16)\t0\t12\t18\t00\t00\t11/Elina 'Konsertissa. En tod. vastaa teille'\tEtu-Töölö, Helsinki (2:40)\t24\t12\t18\t00\t00\t11 I want to parsing the above data into the

Hello list, I have prepared a small tutorial today that deals with how to avoid Asterisk rebuilding DTMF tones when using it to connect industial appliances that use DTMF. You can find it at: http://astrecipes.net/index.php?n=248 I know it isn't everybody's piece of cake, but I thought somebody could be interested as well :) l. -- Home of QueueMetrics -

Does anyone know what the time values in amd.conf are? Are they seconds, fractions of seconds, heartbeats, what? ;'initialSilence' is the maximum silence duration before the greeting initial_silence = 25 ; Maximum silence duration before the greeting. It doesn't say in amd.conf or at http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+AMD --

Hello Sirs, > > Have you ran them through mp3check and mp3_check? > >Yep. Found a couple with a few bad frames and deleted them. Other than >that, nothing worse than a missing ID3 tag or two. Could you please give me the address of these tool(mp3check, mp3_check). I have many bad coded mp3, When Icecast streaming these bad! mp3s it kicks the connected clients. I want to delete

Hello list, I have prepared a couple of new tutorials you may find interesting: - Installing an Asterisk 1.4 beta system - at http://astrecipes.net/?n=216 - Installing the Digium's Asterisk GUI for 1.4 - at http://astrecipes.net/?n=217 It's nothing too complex, but you may find them interesting, especially the new Asterisk GUI. Any comment is welcome - the site is a wiki, so feel

Hi everyone, Installed asterisk from yum repository but I think H.323 is not supported as I tried commands like this and they don't work: - *h.323 debug*: Enable chan_h323 debug - *h.323 gk cycle*: Manually re-register with the Gatekeper - *h.323 hangup*: Manually try to hang up a call - *h.323 no debug*: Disable chan_h323 debug - *h.323 no trace*: Disable H.323 Stack Tracing

Hello, Could the experienced people help me? 1:) Is it possible to start liveice in the background like icecast and shout?? 2:) If I insert another sound card to PC(it will have 2 or more sound cards). Is it enough to specify "SOUND_DEVICE" option to /dev/dspX in liveice.cfg and start another liveice session with that configuration file??? 3:) With many efforts I fail to start darkice

hello, I am tying to run ROR on apache 2 with suse linux 9.3, and I do not succeed with it. I set rubby und rails and all scripts are running fine. my Document root : /srv/rails/demo/public I did not setup FastCGI because I could not run it with normal CGI jet. my Virtual Server runs on 192.168.0.111 ServerName rails DocumentRoot /srv/rails/demo/public <Directory

On Tuesday 18 March 2008 20:17:52 Anton Korobeynikov wrote: > Hello, Everyone > > LLVM recently was approved to take part in Google Summer of Code 2008. > We welcome everyone to apply for this program. > > The list of ideas for (possible) projects is located at > http://llvm.org/OpenProjects.html. Surely you can suggest any other > project, if you feel, that it definitely

Regarding this article (1) I have one question to make. What can I do to record the call if the agent makes a transfer using the "flash" button instead of "transfer button" or using blindxfer or atxfer defined in features. conf If the agent makes the transfer with "flash", the comunication between the person who is calling and is already in the queue and the target

Hello list, I have prepared a small recipe on how to compile Asterisk 1.2 beta 1 with a TDM400 card and H.323. You can find it at http://www.oinko.net/astrecipes/index.php?n=102 Any comment / suggestion / modification /bugfix is welcome! I was wondering: is there any way to build a version of Bristuff for 1.2 beta 1? Bye for now, l. -- Loway Research - Home of QueueMetrics

Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? I've seen this problem consistently with multiple Asterisk installs, 1.2.x and 1.4.x, although admittedly it was more common with 1.2.x as far as I can tell. There is no clearly identifiable sequence of events that causes this to occur, although it