similar to: Dynamic SIP.CONF

Hey all, Quick Question. I have heard mention that Asterisk has the capability to store voicemail inside a database, instead of storing each voicemail in a separate file under a spool directory. Is this true? If so, does it (or can it) use MySQL? Is there any documentation available showing how to do this? The problem that we are having is that we need redundant voicemail servers

Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. --------------------------------------------- patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file

Hi to all!! I'm successful to connect Asterisk to PostgreSQL database... If it's possible, can anyone learn me how to store sip user in PostgreSQL database and how to configure voicemail?? Thanks for all!!!

Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax

Hey All, I've been using Asterisk for a couple years now, but have always had some unsolvable audio problems. I get audio stuttering and popping quite often. Even if I have just one call up! The server is a Dual Duo-Core 3.0ghz Xeon Processor PC with 4 GB or ram. It just seems to me that this should NOT be happening. The server resources are nearly 98% idle. I've tried using

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

hello, I need help with ast_data I downloaded asterisk from cvs cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot co -r HEAD asterisk and the latest ast_data. When i run ./INSTALL.txt i get : serveur1:/opt/asterisk/ast_data# ./INSTALL patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c Hunk #1 succeeded at 27 with

Hi! Today, I found an encoding bug with a new tune by me. http://www.uni-karlsruhe.de/~us87/ogg/vorbis_bassrumble_demo.rar (2.1MB) contains the original .WAV in 16bit/44.1kHz and an .OGG encoded at 350kbit/s. I found the bug when listening to the 128kbit/s version, but encoding it with that high bitrate didn't change a thing. The deep bassdrum contains a rumbling, knocking sound (the first

(Second try to send this. I wonder if the first one gotten eaten by a spam filter; I'll link to patches instead of attaching them.) Here are the tincctl patches I've been working on. They apply to http://www.tinc-vpn.org/svn/tinc/branches/1.1@1545. I intend to commit them once the crypto stuff's fixed. Since they're basically done, I'm emailing them now for review and in case

Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar

Hey All, Is there an app available, or another method, to stream an audio file to a caller while performing additional actions in the background? Regardless of whether DTMF is received or not from the caller. I had originally thought that I could use the "Background" app for this but after further investigation found that Background is primarily for playing audio and waiting for

I'm trying to patch the current asterisk CVS with spandsp-0.0.1k.tar.gz. Everything compiles fine but when I go to patch the asterisk/apps/Makefile it fails: asterisk:/usr/src/spandsp2# patch < Makefile.patch can't find file to patch at input line 3 Perhaps you should have used the -p or --strip option? The text leading up to this was: -------------------------- |--- Makefile.orig

Hi, At first I would like to express how much I like Asterisk. Amazing product. I compiled Asterisk with mySQL support for CDR and Voicemail. Everything seems to be fine, I can see that Asterisk connects to mysql and logs CDRs. I can also see that the VOicemail app is also logged in, however I can not access any mailboxes. Similar messages to others, app_voicemail.c:3011 vm_execmain: Couldn't

Hi After going from AST_DATA (RES_DATA) to realtime with mysql-driver my console is jam'ed with SIP Seeding peers from Astdb '000b8201XXXX' at 000b8201XXXX@81.146.XX.XX:35273 for 120 I got arround 4000 sipclients registered at that server and all the sip-client re-register every 120 sec. so the console is totaly fill'ed with SIP Seeding messages. Is it posible to not see

Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some

I'm still (doggedly) trying to get asterisk to read my voicemail configuration from MySQL. I'm using the stable release of Asterisk, from back in December, before realtime was included. If anyone has got it to work, please contact me ... I've posted details, but everyone who's responded so far has been working with the newer version that uses realtime. Unfortunately, this is

We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 10000 in the future), and I have a few questions:   1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -