similar to: ATA 186 & DynExtenDB (query extensions vía sql)

I'm looking for a way to manage large dial plans. Blitz on IRC mentioned DynExtenDB I'm wondering how stable it is since its not been updated since 2002-12-15 Any other ideas ?? I want to have my dial plan in a SQL database thanks

I have been fooling around with DynExtenDB and run into two glitches. 1) The code is looking for (chan->dnis) and in my case I find (null). I forced (chan-dnis) to be the same as (chan->exten). So far so good. Now I can connect and talk. This lead me to the second glitch. 2) As soon as the call ends by hanging up, the code issues a (ast_spawn_extension). This causes asterisk to drop

On Wed, 30 Apr 2003 00:24:19 -0400, Uriel Carrasquilla wrote: > >Gary: >I just copied the content from chan->exten to chan->dnis. I am calling from How are you doing this coying ? >one extension to another. >Have you got DynExtenDB to work? nope, haven't got over the first problem yet. Gary .

Hi, Am using Asterisk to call peoples phone as part of a service of my website. It will call people for various things...one of them to tell people sports scores. I am using several sound files to piece together a dynamic message saying who played and what the score was. The problem is that I can hardcode the sound files that are needed to play and it works fine, but I cannot hardcode the

Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be

Hi, Does anyone has a *working* patch to set the dnid from extension when it is now supplied, cos this plugin doesn't working without it. I tried and it passes empty string. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/aae0a7f4/attachment.htm

I have at been looking at Andreas Otto's DynExtenDB Module which will forfill half of my needs for a system... The other half requires something similar for registrations in iax.conf Has anyone found a similar solution ?? Gary .

Hi, Can anybody explain me in configuring Asterisk to record a conference? Regards... Girish _________________________________________________________________ Add zing to Hotmail. Get FREE newsletters. http://server1.msn.co.in/features/general/Newsletters/index.asp Subscribe now!

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the "old" way? Thanks in advance, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/6dd07c4b/attachment.htm

Hi all: Somebody knows the mysql table structure for VoiceMail2 application? Thanks in advance, Gus

Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this

How do I use the Authenticate application in my IVR menu, where do I put the password? here is my menu. I need to ask for a password before I let users log into my conference room. [conf1] exten => s,1,Ringing exten => s,2,Wait,2 exten => s,3,Answer exten => s,4,Authenticate(1234) exten => s,5,Hangup exten => a,1,Meetme,1251 I also can not figure out what "Unknown RTP

Hi all: I'm testing a new installation of *, bringing up some ATA186. In * environment, all stuff works greats. The only thing that don't work is a Call Transfer, but the 3Party works ok. Some time ago I read that somebody had proven this functionality successfully. If somebody knows what I missing, please let me know. Thanks in advance, Gus -------------- next part -------------- An

Hi all, I'm using dovecot 2.0.9 and I have a config like this: dovecot.conf: passdb { args = /etc/dovecot/conf.d/passwd-ldap.conf driver = ldap } mail_plugins = $mail_plugins quota plugin { autocreate = INBOX.SPAM autocreate2 = INBOX.NoSPAM autosubscribe = INBOX.SPAM autosubscribe2 = INBOX.NoSPAM quota = maildir } protocol imap { mail_plugins = $mail_plugins imap_quota

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Hi folks, where can I find the R2 beta code for Asterisk? Best, PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030806/9c7a0660/attachment.htm

Hello, I am searching for a solution that I thought should be kind of standard, but until now I was not successful finding anything. Here is the problem: At our site we offer windows and linux, most servers (eg file, samba, web) are linux based. User data is stored on NFS file servers. Windows systems are part of a Windows domain with an ADS domain controller. At the moment the linux samba

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The "show codecs" command shows: *CLI> show codecs 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 <<