similar to: Voicemail to email file sizes

have you set up the db schema? and have you entered any sip data into the db? Sean -----Original Message----- From: Chandra [mailto:chandra@digital.com.np] Sent: Tue 1/6/2004 10:57 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] no results. i have been working with the retrieve_sip_conf_from_mysql.pl file and i have set everything as required. but when i

Hardly finished building our phone system for our school district and I have an opportunity to sell and install a system for a local small business. We are competing against a bid for an integrated voicemail/switch that runs about $1300 (without phones and cabling) and will work with analog phones. Is there hardware configuration (either using analog or IP phones) that would meet these needs and

I just moved my system over to a new server with * 0.7.1. The old machine was using a cvs from August/Sep timeframe. On the new machine I did an make samples but then ovewrote with tar files of the production configs in the /etc/asterisk /var/spool/asterisk /var/lib/asterisk folders. Now the system seems to be working fine but only records blank audio in the voicemail files. Same thing with

you can always do a "restart when convenient" within asterisk, and it will do it's thing when all lines are clear.... -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, January 08, 2004 12:31 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Is there a way to reload a module from the

I have also noticed that sip.conf doesnt get updated without a restart..... was thinking I am doing something wrong, but maybe not now...... Chris -----Original Message----- From: Jonathan Moore [mailto:moorejon@usd465.com] Sent: Thursday, 8 January 2004 8:42 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Re: 911 and lawsuits and redundancy Another concern I have on this

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress on we are having a few disconnects while calls are in session. I have talked both to some local phone contractors and SBC directly and

Hello All, I have a customer site that is using * for ACD. In comming calls are eventually routed to a support rep via a queue. For new accounts the agent needs to be able to send a flash to the PSTN trunk (a POTS line with 3-way calling enabled), dial the number of an authentication center and then connect all three parties together. The trick is that both the agent and the customer need to be

I saw on the wiki that asterisk supports a *0 dial code to flash the external trunk. When I try to use this on my system using a t100p card connected to a channel bank that is agregating 6 POTS lines the code doesn't seem to do anything. Do I need to set a config value somewhere to enable this code? Is anyone using this feature successfully? -- Jonathan Moore Director of Technology Winfield

I just purchased 30 of these after testing one for a few months and would like to quickly purchase another 40. We really like these phones: good sound quality, good echo control (no echo in speaker phone), power over ethernet support, 10/100 switch, 8 programmable keys. Unfortunately we missed on big problem with Call waiting in our testing. When using asterisk 0.9.1 or rc2 the phone will reboot

I just purchased a T100p from digium and a Carrier Access Access Bank 1 channel bank (12fxs/12fxo). I have the setup partially working thanks to some help from IRC. However I still have the following issues I can't seem to resolve 1. When calling into the system from the PSTN call hangup is not detected. * leaves line in use until it is shutdown. 2. When calling an analog phone connected to

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS.

Can anyone help me with the term that SBC uses to refer to disconnect supervision? I have an Adit 600 channel bank which has helped improve the disconnect detection time down to about 8 seconds. This is still causing some issues in particular with call progress enabled in * we are having a few disconnects while calls are in session (about 2 reported in first 5 days of use). I have talked both to

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

Hello all! I am trying to load sip.conf from mysql database. I have followed the instructions at <http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers>. Seems that the authentication (user & psw) works fine but I would like to get more information from mysql and I don't know how to retrieve it. Could anybody help me? Any idea about how to do it? Regards, Victor.

Hi, does anybody know if TMail supports nested multipart messages? I read a changelog that indicates so (http://dev.rubyonrails.org/svn/rails/trunk/actionmailer/CHANGELOG), but I haven''t been able to get it working so far. Maybe I am doing something wrong here? I attached the sample mail. Here is the code to parse it: require File.dirname(__FILE__) +

Hello, I asked a question about what the most likely process to follow if after a time-series fit is performed the residuals are found to be non-normal. One peron responded and offered to help if I supplied a sample data set. Unfortunately now that I have a sample I have lost the emai addressl. If you are that person or have some ideas please email me back at rkevinburton at charter.net. Thank

Hopefully someone can help me. I've been running a windows 2000 Workstation and Server. This spring I added a Linux workstation (Red Hat 7.2 - Samba 2.2.3a) to the network. Took some work but that is running fine. smb.conf seems to be giving me what I want. LAN addresses are assigned with DHCP. I have had several people tell me that they have a network running with static addresses. I

Well, to do an upgrade on a traditional system you have the same issues, perhaps even worse as everything is physically wired to one system. To develop for production you must have a dev environment, a beta test and a scheduled release right? Todd Jonathan Moore <moorejon@usd465.com> wrote: __________ >These are good issues, but I am even thinking of something simpler and more

New Gluster Community Platform: http://community.gluster.org We are pleased to announce a Quora like community platform (beta), where the focus is on *YOU* and your peers in the Community. We wanted to provide a platform for all of you to engage with each other and share everything (ugly, bad and the good) with other community members and us. We want your questions heard and answers recognized.

Heylas! I've been copying mailboxes from one dovecot server (on an alpha running debian testing, 0.99.10-9) to another (on an i686 running debian unstable, 0.99.10-9). I encountered a weird problem, which I seem to have resolved by moving a message with a To: line from hell out of the mailbox. Both machines are running maildir. They're not exactly identical, but they're fairly