similar to: Asterisk 1.2.14 Released

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random -

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.0-beta4. This release contains a number of updates: - a bug fix for the ExternalIVR application and addition of 'silence' sound files to support it - various SIP interoperability improvements - memory and dialog leaks in the SIP channel driver - a fix to music-on-hold random mode that was not really random -

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", "SIP/1014|40|t") in new stack

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

Hello, I recently completed a fresh install of Asterisk SVN-group-srtp-r183146M-/trunk , and I'm running into an issue getting the voicemail application module to load. Output from debug shows: ------------------------------- [Aug 11 22:00:01] NOTICE[20173]: loader.c:875 load_modules: 1 modules will be loaded. [Aug 11 22:00:01] WARNING[20173]: loader.c:376 load_dynamic_module: Error loading

I've opened a few bugs on ExternalIVR and added patches. The biggest issue is: https://issues.asterisk.org/view.php?id=16174 [patch] ExternalIVR does not handle arguments in a consistant manner Basically, this optimizes and fixes several different ways of calling ExternalIVR. If there is anyone who is using ExternalIVR today and/or is willing to test these patches I would appreciate the

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 13.18-cert1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.18-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

Hi all I have been looking at my asterisk CDR in the mysql database and some channel names are set to "**Unknown**" string. When I look at the code, everybody when calling ast_channel_alloc set a channel format like SIP/%s or Zap/%s Only app_voicemail.c doesn't when sending emails and I don't use voicemail. Why app_voicemail needs to allocate a channel to send emails ? And in

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

The Asterisk Development Team has announced the release of Asterisk 12.7.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 12.7.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Bugs

I'm having trouble with ExternalIVR's socket connection. Is it working in the current 1.6.0 trunk? I'm getting this error: Executing [s at ck987_externIVR:1] ExternalIVR("SIP/itp_jnctn-0000006c", "ivr://127.0.0.1:9012") in new stack [Feb 21 13:38:12] NOTICE[15864]: app_externalivr.c:631 eivr_comm: SIP/itp_jnctn-0000006c: stderr: Failed to execute

I have an Perl AGI script that acts as an IVR for my Asterisk box. Basically, it simply plays audio files to the caller, collecting DTMF input and logging every DTMF input into a database table, simply to document every step or option selected by the caller. One thing is that in addition to playing audio files, it also, at some point, plays SayUnitTime command. This IVR has constantly

The Asterisk Development Team would like to announce the release of Asterisk 15.2.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 15.2.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

On 2017-04-18 08:17 PM, Pete Mundy wrote: >> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com >> <mailto:darcy at VybeNetworks.com>> wrote: >> >> <snip> >> Everything looks the same as another one that works except for two >> things. The one that works doesn't have the "Probation passed" lines. >> I am

SOFTWARE FreePBX 2.1.3 CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 Sangoma Wanpipe 2.3.4.5 I had an Asterisk server lock up on me today after 95 days of up time. Had to manually kill the Asterisk process and then restart. Nothing out of the ordinary in terms of memory use as far as I could tell. Seems to be running fine now. Here is the log file. I deleted the stuff in the middle to keep it

I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon <voip@digitaldatabits.net> a ?crit : > These questions should be sent to Asterisk-Users > this is not a

I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon <voip@digitaldatabits.net> a ?crit : > These questions should be sent to Asterisk-Users > this is not a