similar to: Newbie needs help

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report:

If you have many old voicemail messages, to get to the most recent one, you have to keep hitting "6" until you reach the last one. It would be better if you could hit "4" from the first message to get to the last message and/or have a digit that takes you the first and last messages respectively. Anyone have any patches for this?

I am currently having issues when trying to change my password in Voicemail. I am not utilizing realtime because I believe there is a problem with MWI being sent to phones and realtime databases (this may have changed since I last used Asterisk). Whenever I try to change the password for the account, the asterisk program gives a Sig fault (signal 11). I do see the voicemail.conf.new file

Hi All, I have an issue on voice mail with odbc in asterisk 11.7 box. Voice message can be received through Google mail but it doesn't show in phone. The error messages is as follow and let me get your kind advice. -- <SIP/0015-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-00000006]: app_voicemail.c:3824 last_message_index:

I have Asterisk 1.4.32 and I would like save all VoiceMail to MSSQL 2008 database. I configured ODBC, but I receive this errors. [Jun 18 14:00:37] NOTICE[19914] res_odbc.c: Connecting sqlserver [Jun 18 14:00:37] NOTICE[19914] res_odbc.c: res_odbc: Connected to sqlserver [kupalaodbc] [Jun 18 14:00:37] WARNING[19914] app_voicemail.c: SQL Prepare failed![SELECT COUNT(*) FROM pbx_VoiceMail

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hi All, I'd just like to verify what the correct operation of the timeout parameter is for the dial application. I'm not sure if I've encountered a bug or a configuration issue. When a sip phone is not responding to invites on an outbound call, the dial application still waits the duration of timeout before continuing with dialplan execution. I was under the impression that app_dial

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hi list, My need is to append a site specific parameter to the Contact: header on all INVITEs exiting * via a SIP trunk. I'd like it to look something like this: Contact: <bob:3125551212 at 10.10.10.10;SITE-ID=us.here> where SITE-ID=us.here is set in a config file that * parses on startup. Or in a Dial() command option? Or I don't care exactly how. :-) It is possible to

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of voicemail messages into MySQL. It is my understanding that I can only do this via ODBC. I installed per http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to

Hi All, Anyone know if theres a way to share a Cisco 7940 between hot-desk users? My phones get their setup via SIP .cnf files, that load at boot via tftp, so I'm assuming the configs a failry static. However if I want a phone to be hot-desked, I could have different users sitting there. Is there any concept of "logging on" in these environments? Cheers, Adrian

Hi ALL; I got the latest Asterisk-addons for Mysql-Cdr, but I have problem compiling that.It says: # make ..... ........ res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:143: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a cast res_config_mysql.c: In function `realtime_multi_mysql': res_config_mysql.c:242: warning: passing arg

Hi, I need help with modifying cdr_addon_mysql.c I want to have more fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c and replace userfield with ex team (sed -e 's/userfield/team/g' ). When I try to recomplie menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'.