similar to: Asterisk 1.6.0.28 and 1.6.1.20 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.32 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.31 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved with the help

The Asterisk Development Team has announced the release of Asterisk 1.4.31. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.31 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved with the help

The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.8. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.8 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.25. Asterisk 1.4.25 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves several crash issues, DTMF related issues, and CDR related issues. For a summary of the changes in this release, please see the release summary:

The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.25. Asterisk 1.4.25 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves several crash issues, DTMF related issues, and CDR related issues. For a summary of the changes in this release, please see the release summary:

My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a [trunkgroups] [channels] language=en context=incoming toneduration=40 ;usedistinctiveringdetection=yes

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

[18884732963 at from-fax-machine:... - your call is hitting the from-fax-machine context - yet your 'fax' exten is in the from-pstn-4 context. See the "[2011-03-17 13:40:29.6] NOTICE[8825] chan_dahdi.c: Fax detected, but no fax extension" line. When Asterisk detects an incoming fax tone - it tries to automagically route the call to the 'fax' extension in the SAME

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

The Asterisk Development Team has announced the release of libpri 1.4.11.5. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/ The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation. Thank you! The following are some of the issues resolved in this release:

I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS modules. I'm trying to set-up things to route analog fax calls from a FXO port to an analog fax machine on a FXS port on the same card. Outgoing faxes work just fine. But incoming faces are routed to the right DAHDI extension, but the call dropped right as the fax machine rings for the first time. The fax machine

I'm running into a strange issue with Asterisk + Iaxmodem + hylafax on the same machine. After rebooting the iaxmodems don't register to asterisk. Stoping and starting the relevant services gets it working, but what is the point of using init scripts if it does not work right? I already tried to adjust the init scripts in /etc/rc3.d so I have: S50asterisk s90iaxmodem S95hylafax So it

bugzilla-daemon@lists.xensource.com, le Thu 20 Mar 2008 23:23:35 -0700, a écrit : > http://bugzilla.xensource.com/bugzilla/show_bug.cgi?id=1194 > > ------- Comment #3 from haicheng.li@intel.com 2008-03-20 23:23 ------- > This bug may be related to c/s #17266 which says that IDE should accept SETMUL > 0 as upstream qemu now does. It is following: > > ---

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init

Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is "UNKNOW" like: * Name : marco Secret : <Set> Context : phones Parking lot : Mailbox : 2345 at default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk : No Encryption : No

Hi. I've updated sound files in /var/lib/asterisk/sounds and restarted Asterisk, but it still plays old ones. I've removed old files completely then, but still no effect. I've searched for some cahce files, but didn't find any of them. So, could you tell me please, what is an appropriate way to update my sound files? :) Thanks. Asterisk 1.6.1.20

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number

I have an Asterisk 1.6.0.28 system, with a queue called 'marketing'. Everything appears normal, but the status of the members never changes from 'not in use', even if they are being rang or are in a call. Members are added like so: queue add member SIP/1406 to marketing penalty 0 as SIP/1406 state_interface SIP/1406 And they are present as a hint: exten =>