similar to: compile issues.

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

Which version? Similar issues parsing callback number in 1.2.12 > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Kristian Kielhofner > Sent: Thursday, September 28, 2006 10:27 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail callback

List, Having a little trouble with the following. Let me prefix by saying I have blind transfers working from the following setup. Inbound call [from-zap] (SIP/sv0071iv) answers. Zaptel -> Asterisk -> SIP extension SIP extension then blind transfers [from-sip] --- SIP extension -> Asterisk -> Zaptel During this whole process, the original channel off the trunk (lineside T1) is

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]

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:.

Here are a few things I would like to see .. 1. In addition to time/date stamps, store/read the caller id info with the voicemail messages. 2. Have the ability to configure the system to ignore and delete messages left by a caller that are 3 seconds or less (maybe make this configurable) Not sure but that would cut alot of hangup calls out of your voicemail box. I can't think of much more

I'm running the a very recent CVS version of asterisk on an RH9 machine. My problem is that my x100p takes about 10 seconds to detect a hangup. After that it takes about 10 more seconds for the the zaptel device to release the line. Here's an example of my console report: == Parsing '/var/spool/asterisk/voicemail/default/101/INBOX/msg0000.txt': == Parsing

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Following up with some more tests... I just left a message in a voicemail box, and got the same behaviour (i.e. it says its starting to play ('First Message') then it goes to the VM ending menu (i.e. press 7 to delete) without playing the message. On this particular test.. I went ahead & left a 2nd VM in the inbox, then when I went to check them, they would both play properly... The

Hi, After a long way of problems (shipping, customs, etc) finally I got Welltech working. Here below my comments. - The documentation is poor and have errors - The web configuration is not complete. However is useful for the basic configuration parameters. The command line is necessary for modify all parameters. - The software upgrade is easy. Initially the gw came with H323, we upgrade to

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues

The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues

The Asterisk Development Team is pleased to announce the second beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been

The Asterisk Development Team is pleased to announce the second beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been

hello, I tried to compil asterisk-addons-1.6.2.0-rc1, and I have that error: [CC] res_config_mysql.c -> res_config_mysql.o res_config_mysql.c:1367: error: unknown field ?update2_func? specified in initializer res_config_mysql.c: In function ?parse_config?: res_config_mysql.c:1432: error: ?CONFIG_STATUS_FILEMISSING? undeclared (first use in this function) res_config_mysql.c:1432: error:

The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The