similar to: Can one adjust the voicemail-menu when using VoiceMailMain() ?

Hello, I'm running Asterisk 1.6.1.0 compiled from scratch under debian Lenny and I can't delete message from VoiceMailMain using option 7 Default folder is /var/spool/asterisk/voicemail and it's owned by asterisk:asterisk with 777 permissions Apparently VoicemailMain delete the message and inmediatly undelete it ! This the same issue as in this post :

Hello list, using asterisk 1.4.30 and trying GROUP() and GROUP_COUNT() for the first time... Having some troubles. This the dialplan (using a sub) : exten => s,n,Set(_custID=${custID}) exten => s,n,GROUP(${custID}) exten => s,n,NoOp(grouppcount = GROUP_COUNT(${custID})) exten => s,n,GoToIf($[ ${GROUP_COUNT(${custID})} > 2 ]?maxreached) The CLI shows : [Jun 5 16:06:26] --

Hello list, using asterisk 1.4.30 I have the strangest problem that some SIP accounts can register to my Asterisk and others not. I see no connection between all those that can register or all those that can't. It's not a firewall problem as all register to port 5060 and the range 5060 --> 5064 is open. It's just very strange that some can register and other not. Any

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?

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Is there a menu tree diagram somewhere for the Voicemailmain application? I know my users will ask for one, and before I started drawing my own I thought I'd see if someone already had. --- David Brodbeck, System Administrator InterClean Equipment, Inc. 3939 Bestech Drive Suite B Ypsilanti, MI 48197 (734) 975-2967 x221 (734) 975-1646 (fax)

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

I would like to call my own DID number from outside, get into voicemail, and then push '#' to exit into VoicemailMain. Is there a way to do this?

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Hi, I've sent this to asterisk-dev recently, but seen no comments. Has anyone else experienced this behaviour ? I've made a complete clean checkout of CVS code, and it still happens.... > -----Original Message----- > I've just made a new update from cvs on my devel box to play > with, and I noticed that I get console bells when I start the > voicemail app and asterisk

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

Hello list, google returns a discussion on the dev-list when I search for how to mail a voicemail to multiple mail addresses. Is there yet a seperator that actually works to define multiple mail addresses ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I noticed that changes to realtime sip peers are not applied until a 'reload'. A 'sip reload' does not make any changes to realtime sip peers. When changing for instance the mailbox-parameter in the realtime sip_buddies table, the change is not applied with a 'sip reload'. For every change there is a complete 'reload' necessary. Why does a 'sip

It should be as simple as editing voicemail.conf : ; Voicemail Configuration ; [general] ; Formats for writing Voicemail. Note that when using IMAP storage for ; voicemail, only the first format specified will be used. format=wav49|wav|gsm ; Who the e-mail notification should appear to come from serveremail=asterisk-voicemail ; Should the email contain the voicemail as an attachment attach=yes ;

Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, Jonas. -------------- next part

Hello, in asterisk 1.4.30 all realtime configurations go well. In asterisk 1.6.2.11 the following appears on CLI : [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf) [Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql: MySQL RealTime: Invalid database specified: MyDBase (check

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

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:.

Hello list, using asterisk 1.4.30. When setting up the MySQL table 'musiconhold' as described in http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf , what is the meaning of the fields : `*digit*` char(1) NOT NULL default '', `*sort*` varchar(16) NOT NULL default '', and what are there default values ?! What is the default value of :