similar to: No sound into asterisk???

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

I am trying to set up a simple Asterisk server. All it's going to do for now is to act as my voicemail box. I've got a DID from Voicepulse, and am using IAX (I'll get to SIP someday when I want to circumvent the phone company for long-distance, but for now I'd be happy to get a trial version of Asterisk running). So far, I've managed to set up voicemail.conf, extensions.conf

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --------------- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 ---------- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 ----------

Hello all, Per my last message to the list, and my promise to the Developers that I'd create RPMS if they released 0.7.0, I would like to announce the availability of experimental RPMS for Asterisk release 0.7.1. These are targeted at RedHat 7.3 systems, running the latest Kernel release (2.4.20-28.7). As the RPMS mature and people submit comments, changes, updates and patches, I will

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Hello all, It's my birthday today, so as my present I would like everyone possible to download and test my updated set of RPMS for Asterisk 0.7.1. By popular request, I installed and built a set of RPMS for RedHat 9.0, and in the process fixed a bunch of issues from the initial build. I have also updated and will be maintaining a page on the Asterisk Wiki located at:

Currently when I call in to my * box it answers after two rings. I'd like for it to answer without ringing. Is this an option somewhere in the dialplan that I'm missing? Thanks, Andrew

Hi, I've successfully installed Asterisk 1.0 on Fedora Core 2 with the 2.6.8 kernel. I have two other computers running X-lite connecting to it. I've been able to set them up so I can dial extensions "123" and "124" to talk between them. I'm able to access the default "1000", "500", and "600" extensions and they all seem to work.

I hope someone can help me with this. This is what I want to happen. Someone dials in and goes to my extension. First, the phone on my desk rings If there is not an answer, I would like to have the dialplan call my cell phone. If I answer my cell phone, speak the incomming number to me. I press one of the buttons on my cell phone to accept the call. If I don't answer, or I don't

I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

I'm trying to switch to using rsync for updating a huge software library containing binaries, text files, symlinks, and so on. We've been using something homegrown which I'm not that happy with - it's a perl script that systems cp and chmod and such. The problem I'm seeing is: target computer: directories ctime-5 and ctime-5b3 are distinct directories source computer:

Hello. I know that it is recommended to run smbd as a standalone daemon and to avoid inetd. Can you please tell me why inetd is discouraged and what problems it imposes? Also, I have one user who is having problems accessing her personal files on a MacOSX 10.3.2 via smb. Any ideas what may be causing it? Judy Lin NACS-DCS

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

I recently setup app_meetme with the 'i' option. My boss wants users to say their name and go directly into the conference instead of reviewing the recording. If anyone else is interested in this behavior becoming an option, has a suggestion what letter to use as the option (I was thinking 'i' -- with review and 'I' -- without review), or anything else, I'd appreciate

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering

I tried to share my spool directory so I could get monitored calls, and now this error comes up when I try to leave a message in any of my voicemail boxes. Mar 24 12:48:35 WARNING[344081]: app_voicemail.c:1488 leave_voicemail: Error opening text file for o utput -- Recording the message Mar 24 12:48:35 WARNING[344081]: file.c:906 ast_writefile: Unable to open file /var/spool/asterisk/v