similar to: sending AT commands through the SIP channel to the end device?!

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 people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer

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 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, I'm in a bit of a loss at the moment. We have the following situation, we are running Samba for a lot of small companies that need fileservices for there Windows Terminal Servers that they use through a thin client on a Fiber / Lan extention to our datacentre. We have this samba running on 2 linux hosts (Fedora Core 5 and Fedora 7) with a ldap backend for all the domains. This works

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, I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText, because I can just delete the message from my phone (Thomson Speedtouch ST2030) display by sending a return-char (\n). But \n is not escaped: I tried already: exten => 222, n, SendText(\n) exten => 222, n, SendText("\n") exten => 222, n, SendText('\n') exten => 222, n,

Hi, I'm new to the whole R-thing as a replacement for Matlab, not disappointed sofar ;) I found out how to make nice looking boxplots, but i also would like the make a boxplot with 5% and 95% instead of the standard 25 and 75% quantiles. My csv input looks something like: LOCATION FILTER NR DATE VALUE MONTH Peelhorst01 1 14-Jan-94 23.07 1 Peelhorst01 1 28-Jan-94 23.68 1 Peelhorst01 1

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

Hi Ricardo and everybody In this old post to rhelp you say that the problem was solved but not what the sollution was. I have the same problem now. I want to read a Excel file from a google site that has restricted access so I need to give a username and password at some point. I could not find an example or a help file that shows how this can be done. You post is one of few but it does not

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

Hi, I understand SendText() sends text on the current channel. Is there a way to manipulate this feature to SendText toward another SIP device? I use Polycom IP600's. Local sendtext works fine. Would be nice to drop an instant message on another user's phone. thanks! Mark

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

Someone posted a message suggesting someone try sendtext() and so I thought I'd see if it was useful. Much searching through help at the CLI has failed to find any help for sendtext, but I did find that: "core show function vmcount" fails but: "core show function VMCOUNT" works. Is that a bug and if so, has it been reported? Ira