similar to: Asterisk 1.4.25 Now Available

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.0.7. Asterisk 1.6.0.7 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves an issue where IMAP voicemail message retrieval and Message Waiting Indication (MWI) would not work properly with the same mailbox name in multiple voicemail contexts. This release also fixes a

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.1.0. Asterisk 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is the first release in the 1.6.1 branch, which has additional features added since 1.6.0. Please see the CHANGES file for more information about the additional functionality For those upgrading from previous

The Asterisk Development Team is pleased to announce the release of Asterisk 1.6.1.0. Asterisk 1.6.1.0 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is the first release in the 1.6.1 branch, which has additional features added since 1.6.0. Please see the CHANGES file for more information about the additional functionality For those upgrading from previous

I issue this command: RetryDial(another-time,10,4,SIP/GXP280_18,10,ghM(cfmc_dial_private^RetryAndQ ueue^SIP/GXP280_18)) Asterisk rings the phone for 10 seconds. Asterisk then waits 10 seconds. Asterisk rings again for 10 seconds. I would expect this to happen a total of 4 times. The problem is that after the second ring for 10 seconds Asterisk exits the RetryDial step with HANGUPCAUSE=0 and

Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>

I just read: Certain options to the Dial() statement require that Asterisk is in the media path, and consequently Asterisk will not let go of it: /t/, ''T", "h", "H", "w", "W" or "L" (with multiple arguments). Probably there are more. I had in my memory that "r", "R", "m" would also prevent a

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 ? 17:57, John Kiniston a ?crit : > You could try using RetryDial() instead of Dial, It supports playing > an announcement. >

The Asterisk Development Team is pleased to announce the first release candidate 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 community reported issues, including but not limited to: * Backport of state interface changes for app_queue * Fix queue weight behavior so calls in

The Asterisk Development Team is pleased to announce the first release candidate 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 community reported issues, including but not limited to: * Backport of state interface changes for app_queue * Fix queue weight behavior so calls in

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 1.6.2.8. 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 has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to

The Asterisk Development Team has announced the final maintenance releases of Asterisk branches 1.6.0 and 1.6.1 as versions 1.6.0.28 and 1.6.1.20. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The Asterisk releases for 1.6.0.28 and 1.6.1.20 are the last maintenance releases for Asterisk branches 1.6.0 and 1.6.1 and have now moved to

How about: exten => s,1,Dial(SIP/alice&LOCAL/555 at delayed-announce,40) [delayed-announce] exten => 555,1,Wait(20) same => n,Playback(myannouncement,noanswer) same => n,NoOP(Whatever else you want to do goes here) The 'noanswer' option on the Playback means that SIP/alice should continue to ring for the remaining 20 of the 40 seconds, as the Playback will not answer

When the support for h323plus was announced for Asterisk 1.4.25, I tried to build this support in Asterisk. For this, I checked out the h323plus CVS from SourceForge, which reported version 1.20.beta5, and also the ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now I see that I get a segfault whenever I

The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.40 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.4.40. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.40 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 1.6.2.17. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.17 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this