similar to: Variables are empty after Redirecting a channel

> not sure, but the colon in xattr names is probably not handled > correctly at this point. Can you please file a bugreport so we can > track this? Bug 11162 https://bugzilla.samba.org/show_bug.cgi?id=11162 -- HAT

On 06/02/2017 09:43 AM, Martin Kletzander wrote: > [adding back the ML, you probably hit reply instead of reply-all, this > way other people might help if they know more] > > On Fri, Jun 02, 2017 at 08:10:01AM -0400, Michael C. Cambria wrote: >> >> Hi, >> >> libvirtd never seems to get notified that there is work to do. journalct >> -f indicated that

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Hi, i'm trying to implement the following scenario: - A user calls number 700 - Asterisk then dials to extensions 100, 200, 300, 400 and 500 - And then bridges all calls to a conference room I tried to use MeetMe and ChannelRedirect, but seems that after channel redirect nothing more is executed. So, this seem to work for the caller and first called, but the others

Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial

Hi, I am trying to achieve 3-way conferencing taking hint from wiki link http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO Here is the scenario: 1. user "ua1" calls user "ca1" 2. "ua1" then presses the feature code "*0" to redirect "ca1" to conference room 300 3. "ua1" then dials the user "33" 4. user

Hi How to set a threeway conference in asterisk only for VOIP (I am using only SIP channel). Thanks Nikhil

Hello, How to move 2 of 3 users in the MeetMe conference to the newly created MeetMe conference? Dialplan example is welcome. Best, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140123/b64fffc7/attachment.html>

Hi all, I need a command to originate a new channel from dialplan. I should be able to continue execution of the current context after this command. How to do this? Best, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140430/dba42e8b/attachment.html>

Hi ! I'm trying to get a working installation of Xen 4.11 from source, with Libvirt. I managed to compile, install and boot on Xen so far. But when I start the libvirt daemon, it fails with the following error: info : libvirt version: 4.1.0, package: 5.fc28 (Fedora Project, 2018-08-23-19:00:58, buildvm-19 error : libxlDriverConfigNew:1644 : Unable to configure libxl's memory management

Hi, I''ve installed XCP 0.5 and I''m following these steps: http://xenbits.xen.org/xapi/install.html to install ddk VM. But I can''t get network working. I tried to create a vif under xenbr0 then assigned mannually a IP address and to create a vif under xapi0. But in both cases I can ping only dom0 IP. What am I doing wrong? Thanks in advance. -- Sergio Roberto

hi all, I have a little question about meetme in Asterisk. One of my client ask me that all call can, if is necessary, become conference for 3-4 user during conversation. I think that are 2 way for make this: 1- all call (instead if the users are only 2) are conference 2- using n-way call (http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO) I decide to implement the first way because

Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=>1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work.

I'd like to have an extension "join" a call. That is, C can join A and B, just as if it were an analog extension phone. ChanSpy works, sort of. The problem is that once A or B hangs up, the channel is gone. With an analog extension, C would remain connected with B if A hung up. Can I throw A and B into a confbridge and then add C? Create a new channel that grabs the A

https://bugzilla.samba.org/show_bug.cgi?id=10372 Summary: rsync 3.10 error in protocol data stream while rsync 3.0.9 runs through Product: rsync Version: 3.1.0 Platform: x64 OS/Version: Linux Status: NEW Severity: normal Priority: P5 Component: core AssignedTo: wayned at

hi, I hope someone can help me:-) we’ve got a freepbx server. there are 2 special extensions (2001, 2002). if someone calls this extensions (or a call is forwarded to these extensions) and these extension hangup (not the caller party), then we’d like to put the calls back into a queue (1000) and wouldn’t like to hangup. I read your description about hangup hooks:

Hello all, My folks would like to play a message to answering machines automatically after hanging up the phone. So, when the caller dials the number of the callee, hears an answering machine, they would like to enter a code on the phone and hang up. After the hangup the message plays to the callee and disconnects. The message that is played uses text to speech that is tailored to the callee,

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

Hi, I don't know how to call this functionality, but what I want to do is join an already established communication between PSTN---FXS_connected_phone using my SIP phone (I have an asterisk v11 with digium TDM400P at home) Is it possible? What I don't want is using the conference sound and menu.... It's just a normal call between to channels that I have to join for few minutes.

On Sat, Jun 03, 2017 at 09:22:58AM -0400, Michael C Cambria wrote: > > >On 06/02/2017 09:53 AM, Michael C. Cambria wrote: >> >> >> On 06/02/2017 09:43 AM, Martin Kletzander wrote: >>> [adding back the ML, you probably hit reply instead of reply-all, this >>> way other people might help if they know more] >>> >>> On Fri, Jun 02, 2017 at