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Hello all, Is anyone else experiencing trouble accessing svn.digium.com? Neither webview nor svn checkout requests are successful. svnview.digium.com works but svn checkout at this address returns a "repository temporarily relocated" error... Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

The Asterisk.org development team is proud to announce the releases of Asterisk 1.4.22 and 1.6.0. ================================================================= === Asterisk 1.4.22 ============================================= ================================================================= Asterisk 1.4.22 includes a large number of bug fixes for the 1.4 release series of Asterisk. 1.4.22

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

Hello, I have a problem with the meetme application, but I'm not sure if it's a bug or just a misuse. I'm trying to get a 3 way call system working as follow : A calls C B calls C C who's speaking with A or B, presses one keypad (only one) and the 2 incoming SIP (A, B) and C are redirected into a conference room. Therefore, I created an entry in the applicationmap

Hi there, One of our Asterisk 1.2 machine is experiencing problems with MeetMe. Whenever meetme runs, the console is flooded with warning messages: The messages started as "No such file or directory" and becomes "Resource temporarily unavailable". I couldn't figure out what file MeetMe might be looking for, could anyone help? May 4 08:57:38 WARNING[19032]:

Is this me or what? -- Playing 'demo-congrats' -- Executing MeetMe("H323:996", "") in new stack -- Playing 'conf-getconfno' == Parsing '/etc/asterisk/meetme.conf': Found WARNING[17425]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' -- Playing 'conf-getconfno'

Hi, I was switching from zaptel to dahdi and got latest SVN from everything. Compiling works fine. kernel module dahdi_dummy is loaded. /dev/dahdi/pseudo exists Trying to go into a meetme does not work: [Aug 11 14:04:45] -- Executing [8001 at client_int_sgmobile:1] MeetMe("SCCP/6000-00000001", "444|dcIM") in new stack [Aug 11 14:04:45] WARNING[4184]: app_meetme.c:775

Howdy, I am proud to announce that after a good bit of development, community feedback, testing, and code review, the brand new ConfBridge application has been officially merged into Asterisk Trunk!!! http://svnview.digium.com/svn/asterisk?view=revision&revision=314598 If you are already familiar with ConfBridge from Asterisk 1.6.X and 1.8, forget everything you know. This is a completely

In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a !@#$ of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like "Asterisk is single CPU, single core?" I can't access the ILO so I

This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! These branches address various long-standing bugs, most of which are regressions from 1.2. It is hoped that these fixes will solve most of the problems introduced by the

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is pleased to announce the second beta release of Asterisk 10.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ With the release of the Asterisk 10 branch, the preceding '1.' has been removed from the version number per the blog post available at

The Asterisk Development Team is pleased to announce the release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases Asterisk 13 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 11. For more information about support time lines for Asterisk releases, see the

The Asterisk Development Team is pleased to announce the release of Asterisk 13.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases Asterisk 13 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 11. For more information about support time lines for Asterisk releases, see the

Hi gang, Can someone shed some light on the pros and cons of working with the SVN branches of Asterisk vs working in the 1.4 or 1.6 branches? What branch does the SVN release roughly equate to? TIA Danny Nicholas -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I can cross compile asterisk 1.4.21 on arm (imx27) using ltib I want to cross compile the new version 1.6.2.13 but there is an error when I execute the commands : ./configure --build=i686-pc-linux-gnu --host=arm make menuselect The configure seems ok, I have the result info : *configure: Package configured for: configure: OS type : none configure: Host CPU : arm configure:

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also