similar to: sip.conf from mysql

Hello, My question is.. How does cdr_manager work? Does it suppose to populate cdr-csv/Master.csv? What about the cdr table on the database? What is the event some people talk about? I have changed (and reloaded) my configuration of cdr_manager.conf to ; ; Asterisk Call Management CDR ; [general] enabled = yes and it doesn't seem to make any difference. After originate a call from the

Hi, Monitor and soxmix (m option) work fine in CVS Head, not in Asterisk 1.0.9, as the Wiki says. http://www.voip-info.org/tiki-index.php?page=Monitor+setup+sample Anyway I am wondering why asterisk 1.0.9 console shows on Hang up: "monitor executing ( nice -n 19 soxmix "//var/spool/asterisk/monitor/45/47-20050812-113631-in.wav"

Hello, I'm trying to migrate my system to FreeBSD and the Makefile for asterisk-addons fails in the first make clean: bash-2.05b# make clean "Makefile", line 56: Missing dependency operator "Makefile", line 57: Could not find .depend "Makefile", line 58: Need an operator make: fatal errors encountered -- cannot continue I would like to think there is no

Hello, I can understand why asterisk is designed to not to allow two UAs with the same usr/pwd, http://lists.digium.com/pipermail/asterisk-users/2004-February/037284.html, but I have to find a solution for this. My first option is use SER as an extension end of Asterisk, to allow more than one SIP endpoint to register with the same details http://www.voip-info.org/wiki-Asterisk+at+large. I

Hi, I'm trying to transfer an incoming call from the PSTN to another PSTN number through a SER - Asterisk system. SER doing only routing.. pstn call-> SER -> asterisk (call forward) -> SER -> pstn Logic for SER: If something comes from the pstn, send it to asterisk. If something comes from asterisk, send it to the pstn. Every time I am getting a "Got SIP response 481

* --> SIP --> CISCO --> PRI --> PSTN The PSTN sees no callerid. *---> PRI[zaptel]--> PSTN Callerid is there... which makes me think it's the cisco, not the PRI/PSTN/telco. CISCO PRI--> * PRI [zaptel] Callerid IS there... which makes me shake my head in disbelief, because * can see clid from the cisco pri, but pstn doesn't... but when * sends info on that pri,

Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? Thanks for your help... I've read voip-info, and various other sources, and search engines, and google...

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.

Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well as a lack of spy/whispering commands available via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor

Hi samba members, I've searched through google, the archives and RTFM but I can't figure this out. I've got a setup of two different offices with both their own samba PDC, wins, etc. Via OpenVPN i've set up an interdomain trust between the two domains (for now one way, but this will become two ways). One domain is called PSW with bluemoon as PDC, the other PSWINDWG with redmoon as

Hi, I'm trying to use the spandsp fax-back facility and despite I can send and receive faxes, this is not working fine. I would like to get a debug of what is going on. I am using the flag debug, but I don't know if txfax is generating any log information or where it is saving it. I just don't find anything. My line in extensions.conf is: exten =>

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

Hi, How could I change the default permissions for voicemails? When I try to install the patch mentioned at http://www.voip-info.org/tiki-index.php?page=Asterisk+gui+vmail.cgi, I get the following response: patch < voicemail.patch patching file app_voicemail.c Hunk #1 FAILED at 39. Hunk #2 FAILED at 119. Hunk #3 FAILED at 296. Hunk #4 FAILED at 1248. Hunk #5 FAILED at 1273. Hunk #6

here's how i mounted a shared folder: #mount -t smbfs -o username=me,password=psw,workgoupt=kppo //har-file2/new-gp /samba/k-drive and when trying to un-mount using umount /samba/k-drive or, smbumount /samba/k-drive I get this message: 'device is resource busy' Any idea why i can' unmount this? please reply to all, thanks!

Hola a todos. Estoy atascado en un sitio que no me he encontrado antes, debe ser fácil y rápido, a ver si alguien me puede ayudar. Puesto fácil, tengo: > aa <- matrix(11:19,3,3) > aa [,1] [,2] [,3] [1,] 11 14 17 [2,] 12 15 18 [3,] 13 16 19 Quiero la fila cuyo primer elemento es 12 y el segundo 15. > aa[aa[,1:2] == (c(12,15)),] Error: (subscript) logical

Hello, I'm trying to install CAPI Driver for Suse 9.2 and I found the documentation for this pretty old since It refers to Suse 8.2 ( http://www.voip-info.org/wiki-Asterisk+AVM+Fritz+CAPI+Driver+Install ). This is especially apparent when I look at the section of these instructions for altering "src.drv/makefile" to replace the occurance of "CARD_PATH". I tried to

It's pretty obvious from the wiki that realtime and Nat don't befriend quite well. As It is obvious the necesity of both of them, mainly have clients under nat talking to an asterisk server. The question I would like to throw away is.. What would you do to have both of them? I have two possible solutions in mind. 1. Use static configuration for sip users with nat=yes. 2. Buy iax

On page 22 of the R-introduction guide it's written: the quadratic form x^{'} A^{-1} x which is used in multivariate computations, should be computed by something like x%*%solve(A,x), rather than computing the inverse of A. Why isn't it good to compute t(x) %*% solve(A) %*% x? Thanks a lot for help!

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

Hi I migrated to dovecot 2 and I started using lmtp for mail delivery. But I have found this on my logs "save failed to INBOX: Not enough disk space" . There is enougth space on disk. and I am not using quota plugin on lmtp my lmtp conf is as follow: # ## LMTP specific settings ## # Support proxying to other LMTP/SMTP servers by performing passdb lookups. #lmtp_proxy = no # When