similar to: make script 1.6.0.6 breaks up, need help!

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, Anyone knows if TRANSFER event on queue_log is still working on 1.6.0.6. I make an attended transfer (asterisk feature), and I cant see the event. Any idea? Should I submit a bug report? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090313/eb5a7ea0/attachment.htm

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

I'm working on a project that involves Queues with Agents that are at home with a PSTN phone number, NOT connected via SIP phones. In the queues.conf it clearly states that only the SIP driver supports "In Use" detection of making members of a Queue available or unavailable. I've not yet figured out the best way to handle this. Currently I've got a macro that is executed

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 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, does anyone know if there is an equivalent dial option for the source channel to the g option? I've had a good look and can't find one. g - Proceed with dialplan execution at the current extension if the destination channel hangs up. Thanks Lee -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have not used Asterisk for some time, but decieed to have a go with it again. I noticed that some commands have been changed, where can one find a list of them except for the help command? I want to simulate a phone like I could do in previous versions of Asterisk so i can type dial and an extension from the Asterisk CLI. Many thanks, Christian

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! 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

cdr_mysql doesn't set the userfield when it's set inside a macro called from a feature (1.4.25, addons 1.4.8). I have a feature code: autorecord => *1,self,Macro,apprecord The apprecord macro looks like: [macro-apprecord] exten => s,1,Playback(beep) exten =>

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

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

Rob Burris wrote: > Real Networks offers a free streamer called Real Server. I'm not sure of the > exact link, but here's a starting point That is definitely not free, it's not even zero cost after that evaluation year. There don't even exist any players for it except from Real ... i can't think of anything less free than that. :| I suggest ditching all that crap and to

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

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.