similar to: Asterisk-Addons 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1 Now Available

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and 1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

Hi all, I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording. I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key. The problems is that, Asterisk

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

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

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These

Dear All, i have following CLI error while try to run this command from Dialplan *TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute 'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into

Hi List. I have an IP Phone when I'm on a call and tightness in the Transfer button, it opens a new channel for me to make a new connection. But the extension is on hold is that mute the music without the wait. How should I proceed to solve my problem. I'm using asterisk 1.6.0.5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

I first asked this question at the end of October, got a response which didn't solve the problem, and have finally got back to the issue now. I'm running 2.2.7a, recently upgraded from 2.2.0 as Domain Logons for Win2K were suspected to be part of my problem. Anyway, the issue relates to AutoCAD (actually a vertical product called Land3) and a command which works with winXX boxes, but

On 4/7/20 11:54 AM, Gary Stainburn wrote: > I have fail2ban on my mail server monitoring Dovecot and Exim. > > I have noticed that it has stopped banning IP's. I have seen in /var/log/fail2ban.log: > > 2020-04-07 09:42:05,875 fail2ban.filter [16138]: INFO [dovecot] Found 77.40.61.224 - 2020-04-07 09:42:05 > 2020-04-07 09:42:06,408 fail2ban.actions [16138]:

I have fail2ban on my mail server monitoring Dovecot and Exim. I have noticed that it has stopped banning IP's. I have seen in /var/log/fail2ban.log: 2020-04-07 09:42:05,875 fail2ban.filter [16138]: INFO [dovecot] Found 77.40.61.224 - 2020-04-07 09:42:05 2020-04-07 09:42:06,408 fail2ban.actions [16138]: NOTICE [dovecot] Ban 77.40.61.224 2020-04-07 09:42:06,981

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone

The Asterisk Development Team has announced the release of Asterisk 1.6.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.1 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * CLI 'queue show' formatting fix. (Closes issue

The Asterisk Development Team has announced the release of Asterisk 1.6.2.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.1 resolved several issues reported by the community, and would have not been possible without your participation. Thank you! * CLI 'queue show' formatting fix. (Closes issue