similar to: Asterisk 10.11.0 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.11.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 10.11.0 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 11.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.1.0 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 11.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.1.0 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 1.8.19.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.19.0 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 11.0.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.0.1 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: * ---

Hi All, We have a requirement to record over 60 simultaneous calls. Our recording facilities are implemented using Monitor() over AMI. The thing we have noticed that making 60 simultaneous call recordings using wav CPU load is significantly higher (around 2 times more) than using gsm. Even writing call recordings to /dev/null makes a big difference in CPU load. What could be the reason for this?

For some reason we are seeing "Avoiding deadlock for channel" in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are "Avoiding deadlock". What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's.

https://bugzilla.samba.org/show_bug.cgi?id=3491 Summary: throttle disk IO during filelist/directory parsing Product: rsync Version: 2.6.4 Platform: All URL: http://vilius.multiply.com/video/item/10 OS/Version: Linux Status: NEW Severity: enhancement Priority: P3 Component: core

We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same =>

Hello, Anyone see this issue, I have a conference bridge setup for a church with a Barix unit that streams audio into the bridge. The bridge is started by calling in to a number that executes a call file and the system calls the Barix unit starting the broadcast. Users then call in and can listen to the sermons live. The system works flawless with 1 issue I can't get accurate cdr's. Every

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBridge:81 ;;exten => 82,hint,MeetMe:82 exten => 82,hint,ConfBridge:82 ;;exten

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file "conf-kicked" and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-00000000) 2] Normal User (e.g. SIP/8484-00000001) 3] Admin User (e.g.

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the conference, he does not become the single video source of the conf. The video mode stays follow_talker. I

To all: No matter what I try, I cannot get the system to wait for the admin to join. It just dumps users into the bridge directly. I do not have a pin for users, does that matter? What am I missing? Another issue the absolute timeout is not working ? ... have recordings that last for over 24 hours... and this should not happen... All calls should hangup after 4 ? Any ideas ? Any help is much

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

Hi guys How can I record the confbridge only when after a marked user is logged in to conference ? Is there any option on the confbridge to start recording when marked user is logged in instead of when the first user logs in ? I tried setting up "same => n,Set(CONFBRIDGE(bridge,record_conference)=yes)" before sending marked user to the conference but it won't start the recording

i am currently not in singapore from the 6th to 10th Aug 2008. For anything urgent, please contact Daren at daren@hwzcorp.com Thanks. Best Regards, Choon Kiat _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

i am currently not in singapore from the 6th to 10th Aug 2008. For anything urgent, please contact Daren at daren@hwzcorp.com Thanks. Best Regards, Choon Kiat _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users