similar to: asterisk-GUI

Hello, I'm novice on the SIP protocol also on Asterisk. Could someone explain me why the Asterisk is using username and secret config on peer connection? Does Asterisk can send call to peer without username and secret configuration ? Sincerely, Tseveen.

Hello, I just installed asterisk 1.6.1.1, asterisk-addons 1.6.1.1 and Asterisk-GUI 2.0 on Ubuntu 9.04 from *source. *Asterisk-GUI's web interface doesn't appear from this URL http://localhost:8088/asterisk/static/config/index.html*. *It says Not Found The requested URL was not found on this server. ------------------------------ Asterisk Server I already defined following config *in

Dear All, I've installed Postfix 2.7.0 and Dovecot 1.2.9 on Ubuntu 10.04. I want to use Dovecot SASL with Postfix and did the following configuration. main.cf smtpd_sasl_auth_enable = yes smtpd_sasl_type = dovecot smtpd_sasl_path = private/dovecot-auth smtpd_sasl_authenticated_header = yes smtpd_sasl_security_options = noanonymous smtpd_sasl_local_domain = $myhostname

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, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

Hi, Reading http://www.asteriskguru.com/tutorials/bri.html , it seems PtP is the way to connect businesses but if you read http://public.swbell.net/ISDN/connect.html you would think the opposite. Can anyone elaborate a bit PtP or PtmP respective advantages ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i tried to compile Asterisk 1.4.21.2 on a server which i have been using with many previous Asterisk versions, without any problems. But with 1.4.21.2 it failed: ---------------------------------- [CC] res_adsi.c -> res_adsi.o [LD] res_adsi.o -> res_adsi.so [CC] res_agi.c -> res_agi.o [LD] res_agi.o -> res_agi.so [CC] res_clioriginate.c -> res_clioriginate.o

Hi! Asterisk in Debian/Lenny claims to be bristuffed, not? At least the the Debian patch tracking system shows the bristuff-patches: [1] http://bit.ly/bRRHe7 We have a QuadBRI-Card and recently needed support from Junghanns.net but they refused telling us there is no bristuff installed because of the show version output: *CLI> show version Asterisk 1.4.21.2~dfsg-3+lenny1 built by pbuilder @

Hello, looks like Asterisk (at least the Version 1.4.21.2 from debian lenny, which I use) has a serious Problem with pseudo-realtime mode (-p switch) in conjunction with recent Kernels. Whenn using -p Asterisk simply hangs forever on startup. This is almost certainly related to the CFS SCHED_FIFO changes: http://lwn.net/Articles/296419/ Do you know how a workaround or patch how to run asterisk

Hi guys.. I just wanted to know if this config could work correctly, since a lot of u guys have been working a lot with asterisk... Asterisk 1.4.21.2 -> chosen because is the last with Zaptel support ? Zaptel 1.4.12 Add ons 1.4.9 Currently i have a 1.6.1.0 Server running with no problems, but I'm about to buy a openvox a400p Card, so i would like to test both environments, 1.4 and 1.6,

Greetings, This may be a dumb question, but here goes. When I was on 1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to line supervision on my POTS lines using a TDM400P/TDM410P. Since upgrading to the DAHDI branches of 1.4 (SVN and 1.4.26.1), I've only been able to duplicate the success of the 1.4.21 functionality once. To test what I'm talking

Hi Everyone, I have two servers as the following that are trunked with each other via IAX2 trunk: Server A: Asterisk 1.4.21.2 (Elastix Flavor) Server B (IP # 72.72.72.72): Asterisk 1.6.2.0 (Vanilla) Server B can place calls to Server A but when trying to place calls from Server A to Server B this is what I am getting:

Hello asterisk users... We are having a little problem in our installation, we are using Asterisk 1.4.21.2 and zaptel 1.4.11 with a Digium TDM410P (3FXO + 1FXS), the problem is that when we disconnect the line from any of the fxo ports, we receive an incoming ghost call (using zap/x channel) it rings on the phone but we cant hear nothing...it's always doing the same everytime we disconnect

Hi all, I have a demo machine I'm running up on Lenny - it has the packaged Asterisk version installed (1.4.21.2+stuff). I'm trying to add an extension to leave a voicemail message, just with Voicemail(1234), which I've done before (on 1.2 at least), but it's saying "no application 'Voicemail' ". "module show like voi" shows

Hello, I am trying to get a repeatable build setup for asterisk. Part of doing so involves using the -C option to specify the master config file. The problem is that asterisk reads the config file location that i specify, however it still tries to read two other config files, namely: * /etc/asterisk/extconfig.conf, and * /etc/asterisk/logger.conf I have specified in my config file that the

Hello, all. I am delightfully slogging my way through installing and configuring Asterisk 1.6.1.1 on CentOS 5.3. I'm learning lots and admiring the product but I'm having a problem getting speex to install and I would very much like to use it. It is not available in menuselect and the problem appears to be with speex_preprocess_ctl: [root at pbx01 asterisk-1.6.1.1]# grep -i speex

Hello About every three months, my dad's little Asterisk server that handles his business phone line with an OpenVox PCI card stops taking calls. To check if it's the cause, I'd like to run a CRON job every night to restart Zaptel and Asterisk. Before I go ahead, I'd like to know if I can just send the following commands, or if there are issues I should know about:

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

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

Hello, I can't get the sound over alsa to work with Asterisk. My current version is 1.4.21.2~dfsg-3 running on debian stable. All settings are the default ones with exception of: /etc/asterisk/modules.conf: load => chan_alsa.so noload => chan_oss.so /etc/asterisk/alsa.conf: input_device=default output_device=default asterisk is started up and doesn't complain about alsa in