similar to: Asterisk 1.6.0.5 no MusicHold REFER

Asterisk supports SIP-T? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090217/7cc67800/attachment.htm

Boa Tarde Lista. I'm having problems in tramiss?o a fax using T.38. My scenario is: Asterisk 1.6.0.5 2 ATA of Intelbras 2210. ReceiveFAX in the asterisk. Unable to fax when it is a ATA to another user on the Asterisk means, if I directly between the ATA works perfectly, is a step to the ATA ReceiveFAX of Asterisk works perfect, but if I try to pass between two Branches

Hi List. It is possible to travel over the X.25 protocol on Asterisk SIP? -- Atenciosamente Daviramos Roussenq Fortunato -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100928/7d6ca5fd/attachment.htm

Hi List. I have a small problem in using the transfer key transfer of IP Phone in Asterisk 1.6, I think I spend some detail in the configuration but can not find. What happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when

Hi I am using asterisk 1.6.0.5. I have one conference say 1234786 and in this conference 25 users are talking with each other.. In this 25 users, 5 is admin/marked and 20 are normal.. Admin user has rights to mute/unmute all user by executing action: meetmemuteall with meetme number. While executing MeetmeMuteAll action, this action will mute all 20 normal users but not admin.. This thing work

Hi, all. I've been searching google, bug reports and forums and have looked in all the asterisk-users list archives back to 2003 but haven't seen an answer to this, so thought I'd post here. The problem seems to be that Asterisk 1.6.0.5 is sending backslashes (needed to escape commas and so forth in 1.4.21.2) as *literal* backslashes to Mysql, so that Mysql gives a syntax error

While working on the mge-hid subdriver, I wanted to add the 'beeper.mute' command. Unfortunately, the description I intended to use was used already by the 'beeper.off' instcmd: CMDDESC beeper.off "Temporarily mute the UPS beeper" Now of course we could add another instcmd 'beeper.reallyoff', but instead I would prefer to bite the buller and do the following:

Hi all, Trying to upgrade a system from 9.0-RELEASE to 9.1-PRE from yesterday on my machine (GEOM+ZFS mirror setup on ada[01]p3), the new kernel becomes unable to mount root... The only way to recover is to boot from 9.0 kernel. The disks were already named ada[01] in 9.0, so I suspect nothing there... I tried - disabling AHCI in bios (no change seen) - change cables, check PSU, test disks

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

The Asterisk Development Team has announced releases of Asterisk-Addons version 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve several issues reported by the community: * Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream. (Closes issue #15224.

The Asterisk Development Team has announced releases of Asterisk-Addons version 1.4.11, 1.6.0.5, 1.6.1.3, and 1.6.2.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ These releases resolve several issues reported by the community: * Fix reading samples from format_mp3 after ast_seekstream/ast_tellstream. (Closes issue #15224.

Daniel Ballenger wrote: > Easy, open up the volume control panel and turn the line in down/mute it I wouldn't fiddle with the fader as some cards use the same setting for playback and record level. But you should be able to mute it just fine by checking the mute checkbox for the line in. Geoff.

The Asterisk.org development team has announced the release of Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5. These releases are available for immediate download from http://downloads.digium.com/. This update for Asterisk includes a security fix for chan_iax2. Please see the associated security adivisory for more details: http://downloads.digium.com/pub/security/AST-2009-001.html These

I tried using the iLBC codec, and whlie I like it, I ran into a strange issue. I did a few searches on Google and haven't found anyone with the same issue as this. Anyhow, I was using a Plantronics analog headset and box plugged into a Digium TDM card, dialed out over my VoIP provider's IAX channel to the PSTN. I was in a conference call which is running on an Avaya PBX (which

Hi I'm having a very odd phenomenon happening on our production server (1.4.17 and using realtime). Sometimes a call will disconnect 30 seconds after the SIP phone hits the mute button but it doesn't happen all the time. I've done a sip debug while watching this happen and that doesn't show anything other than a BYE message being sent out of the blue. The rtptimeout and

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

Dear R-helpers, I am using confint() within a function, and I want to turn off the message it prints: x <- rnorm(100) y <- x^1.1+rnorm(100) nlsfit <- nls(y ~ g0*x^g1, start=list(g0=1,g1=1)) > confint(nlsfit) Waiting for profiling to be done... 2.5% 97.5% g0 0.4484198 1.143761 g1 1.0380479 2.370057 I cannot find any way to turn off 'Waiting for. .." I tried

I have a CyberPower PR3000LCDRTXL2U with a BP48V75ART2U expansion chassis, which I am monitoring using NUT 2.8.0 (on Gentoo Linux). TThe UPS appears to be telling me that the batteries need replacement due to age. CyberPower support has confirmed that for me and told me how i should be able to mute the alarm from the front panel until I can replace the batteries, but it doesn't appear

Hi all, I use an APC Back-UPS RS 800, overall my experience with NUT is very good but there is one feature I am missing: the ability to ask the UPS for a battery test. upscmd -l shows the following commands available: $ upscmd -l rs800 Instant commands supported on UPS [rs800]: beeper.disable - Disable the UPS beeper beeper.enable - Enable the UPS beeper beeper.mute - Temporarily mute the UPS

Dear All, i have following CLI error while try to run this command from Dialplan *TrySystem("DAHDI/45-1", "asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into dundilookup"") in new stack WARNING[32626]: app_system.c:81 system_exec_helper: Unable to execute 'asterisk -rx "dialplan add extension 1234111,1,Goto(incomingdundi,s,1) into