similar to: app_dbodbc for asterisk stable 1.09

I have been searching for the last two days and I cannot seem to set Asterisk to work from a database, can someone please tell me what I am doing wrong here? Here are my files [app_dbodbc.so] => (Database access functions for Asterisk extension logic) == Parsing '/etc/asterisk/odbc.conf': Found > app_dbodbc: dsn is MySQL-asterisk > app_dbodbc: username is

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

Hi I am trying to use SetCDRUserField in an agi script but with no success. I am using the CDR mysql addon, however I can't see it being at fault as my attempt is not doing anything to the CVS CD either. has anyone used this, any hints guidence would be greatly appreciated. The syntax I am using is like so .. res=DoExec('SetCDRUserField','12345'); and then dialing the

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi Guys, I am having trouble trying to get format_mp3 working to play music on hold. I have followed the instructions in the read-me and the wiki however it seems after un-installing mpg123, asterisk is not even attempting to play MOH. My musiconhold.conf is ; Music on hold class definitions ; [classes] [moh_files] default = >/var/lib/asterisk/moh-native ;default =>

Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am

Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

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

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

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

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

I have done a little work on asterisk and database integration. Below is a link to app_dbmysql, modeled after Brian's app_dbodbc but for pure MySQL. I also ported the mysql-vm-routines.h to ODBC in case anyone is interested. You can get both of these from: http://www.cheapnet.net/~mike/asterisk They were working as of yesterday CVS, but today CVS will not compile and I have not looked

The 9p-virtio transport does zero copy on things larger than 1024 bytes in size. It accomplishes this by returning the physical addresses of pages to the virtio-pci device. At present, the translation is usually a bit shift. However, that approach produces an invalid page address when we read/write to vmalloc buffers, such as those used for Linux kernle modules. This causes QEMU to die printing:

The 9p-virtio transport does zero copy on things larger than 1024 bytes in size. It accomplishes this by returning the physical addresses of pages to the virtio-pci device. At present, the translation is usually a bit shift. However, that approach produces an invalid page address when we read/write to vmalloc buffers, such as those used for Linux kernle modules. This causes QEMU to die printing:

Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as

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

Has anyone tried using BroadVoice for VSP? I have Asterisk configured for a home office & I've been trying to decide which VoIP provider to go with for a little while now. I had heard you could get sub $.01 calls but I have not found that to be true yet (not saying it's not possible, I just haven't found it!). Also I'm not sure if BV will support multiple lines. Any

[New Features] 1. Added focus and tab-order to all input fields 2. Dynamic generation of date/month/year listboxes a. It is no longer possible to schedule an invalid date. 3. Added 'Extend' and 'End Now' buttons to the monitor page. 4. Invite button on the monitor page. This greatly simplifies the process of adding callers to a conference. The ./lib/defines

Hi, though this is a bit of a side question, has anybody had an issue while running Outlook 2010 with Dovecot? The reason why I am asking is that I have setup a Dovecot 2.1.7 server on FreeBSD which works fantastically with Thunderbird but Outlook seems to be twice as slow in transferring information across?? # dovecot -n # 2.1.7: /usr/local/etc/dovecot/dovecot.conf # OS: FreeBSD 8.2-RELEASE