similar to: Dial() cmd executing Macro - dropped audio

Hi All, suppose this: Dial(SIP/<somecarrier>/<somenumber>/60/L(3600000)M(td|${EPOCH}) where 60 is the seconds to wait for the callee (the called party) to answer L(3600000) is the absolute limit of the call once it has been answered, in ms M(td|${EPOCH}) is the macro to execute when the call gets answered. ${EPOCH} contains the current unixtime. That's the macro: [macro-td]

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

> On further investigation, the SMP affinity on ALL of the IRQs is set to > 00000001. This implies that everything is handled by CPU0, which it clearly > is not! > > I'll do some more research on this but in the meantime if anyone has any > advice on this issue I would appreciate it. There is a kernel bug in Fedora flavors < 2.6.15 that prevents SMP affinity from being

Used the Directory application instead of the Directory AGI. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option

I subscribed to this list for about two months before I began posting, so I've got a buttload of email to sift through ... I'm doing this BEFORE I flood the list with my inane questions ... But here goes: I read a reply from one guy to another about recording. The message included this context from extensions.conf: [recordings] exten => 500,1,Festival('Please record your

<rant> So, we've got a couple of WinDoze servers, and we getting logs from them. Except the logs seem to stop on occasion. My manager set me onto this (I *loathe* WinDoze, but someone's gotta do it), and I finally found out that a previous co-worker installed eventlog-to-syslog, which seems to be hosted in google code. And it *seems* to have come from Purdue Univ. So < found the

Hello: Thanks for the input so far. Heres the issue-- This is a production environment-- where many people "touch" the files. ie-- The audio engineer is a freelancer who wants to master the files at the highest quality TO HIS EAR and experience-- He knows NADA, Not a thing about SOX-- but is a ProTools GURU. The SOX resampled files work on our asterisk box-- but I gotta put someone

Hello, I'm pretty new to asterisk (only been fighting with it on and off for about the last month), so please go easy. I've been wrestling with the documentation, forum posts, google, and my lack of telephony and VOIP knowledge, trying to get my setup to work. My current problem has me stumped saying, "There's gotta be a cleaner way to do this." Please show me the light

Hello Everyone, I gotta problem that effects 2 of my products. I was wandering if I could get your take on it. Basically, I have 2 parallel effects. First Effect makes things disappear new Effect.Parallel( [ new Effect.Fade(oTab.panel, {duration: 1}), new Effect.Fade(contentId, {duration:1}), new Effect.SlideUp(this.panelContainer, {duration:1}) ], {queue:{position: end, scope:

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for

On Wed, May 6, 2015 3:26 pm, m.roth at 5-cent.us wrote: > <rant> > So, we've got a couple of WinDoze servers, and we getting logs from them. > Except the logs seem to stop on occasion. My manager set me onto this (I > *loathe* WinDoze, but someone's gotta do it), and I finally found out that > a previous co-worker installed eventlog-to-syslog, which seems to be >

Alright, I'm trying this message again. I'm receiving tons of traffic on the Vorbis list, but so far my submissions have been silently rejected. I'm very concerned about a serious usability feature lacking from the current MP3 spec, and possibly from Ogg Vorbis. It has nothing to do with bitrate, but everything to do with quality. I listen to progressive rock and live albums quite

AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc:

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

I understand that standalone macros have been deprecated in 1.6 for gosub routines. I've been working on converting them all but was wondering about dial macros - it doesn't look like there's a replacement yet to call a gosub routine from the dial command. Or am I looking at this wrong? hose

First of all, this is my first post so please go easy if I'm a little off-topic. Just let me know nicely off-list. Thanks! As Mike already knows, I am currently trying to develop a plug-in for playing Shorten files over Audion on the Macintosh platform. When I proposed this to him to get some information on contacting the right people, he mentioned you guys and I need a little info.

Thanks to help from Jim Dickenson I managed to start a macro and get info about the channel that picked up the call from my ruby script. The only thing that I cant do so far, is capturing the ${CHANNEL} variable in the ruby script that started the macro. Is that variable accessible from the ruby script too or just from the macro? Here's a snippet from my ruby script:

Hi, How do I find out the extension that answered an incoming call from inside a Dial(number,M()) macro? I've tried MACRO_EXTEN but it seems to be empty. exten => s,100,Dial(${EXTENSIONS},40,M(support)) [macro-support] exten => s,1,NoOp(${MACRO_EXTEN}) etc... Thanks in Advance, Adam.