similar to: Don't get asterisk to run behind NAT router

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! 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 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 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 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

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

Sorry for the crosspost. This was supposed to be on -users I know some of you are polycom gurus... Anyone know how to remove transfer from a polycom 33x phone? We've set allowtransfer=no, but we would like to remove a polycom soft key as well.

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

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

Hi All, I need to stop the transfer feature on particular sip user. I am using linksys phone and it has set the forwarding enable to another user. I have three users 2101, 2102, 2103. 2102 is registered in linksys phone with forwarding enable to 2103. But is there any procedure in asterisk that we can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved

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, 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 can be useful. Our common requirement for student is to submit proposal to LLVM Developers

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

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI> show hints -= Registered Asterisk Dial Plan Hints =- 30@default : State:Unavailable Watchers 3 29@default :