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timeout to be set. I'm hoping to find an option along the lines of the Dial() ringtime, but no luck. Gosub() looked interesting, but I don't think quite fits my needs either Could someone please offer a little insight on this situation and point me towards the right command to be playing with? [1112221234] exten => s,1,Ringing exten => s,2,Wait(1) exten => s,3,Answer exten =>

Hello all, This is a VICIDial server and I am looking to send calls to VM box 2100 after 3 minutes of sitting in the queue(via the VICIDial AGI). This would be inserted between exten => s,8,Background(open) and exten => s,9,AGI.

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

"Your server has unexpectedly terminated the connection..." "...POP3, Server Response: '+OK 795 octets'..." Interestingly 795 bytes is the size of the "bad" email (the one that won't download). Any thoughts? Cheers, </pre> <blockquote type="cite"> <blockquote type="cite"> <blockquote

penSuse 12.1. Lets check with OpenSuse 12.1.&nbsp; <div><br /> </div> <div>Regards.</div> <div><br /> <br /> <div class=3D"gmail_quote">On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan = N <span dir=3D"ltr">&lt;<a href=3D"mailto:gopalakrishnan.an at gmail.com" targ=

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

If you don't use 'CERTVERIFY 1', then this will at least make sure that nobody can sniff your sessions without a large effort (...) > So, do I misunderstand CERTVERIFY directive ? Or is there a bug ? >> Can you reproduce such behaviour ? >> > > I'm not sure what is going on. Can you try running 'upsmon' with debugging > enabled? The following are

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

Or even: <a class="moz-txt-link-freetext" href="http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946">http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&amp;mid=4946</a> (same thing from the UK site:) <a class="moz-txt-link-freetext"

Hello List! I have 2 asterisk servers, The Admin(.20), and the Call Center(.21). The Admin server contains the 1XXX extension and the Call Center hosts the 2XXX extensions. I would like for our Admin folks to be able to call the Call Center folks (and vice versa). The call will go over the server fine, but when the Call Center server answer, the CLI returns: "NOTICE[4296]: chan_iax2.c:7398

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

We are using G711 for phones to talk to Asterisk and G729 licenses at asterisk to talk to ITSP Could you please suggest transcoder to use from G711 and G729 and which is comptible with Asterisk. We will like to avoid using TDM if possible Also i remember that initially we didn't have G729 and were using only 711 for with vicidial but then also we had same problems. at that time it was only 2

</font> <ul class="D"> <li class="D_off">Two Available PCI Express x8 Slots</li> <li class="D_on">Two Available PCI Express x8 Low Profile Slots</li> <li class="D_off">One Available 64-bit/100MHz PCI-X slot</li> </ul> The list has already answered what goes in what slot so I won't repeat that.

--============_-1179735293==_ma============ Content-Type: text/plain; charset="us-ascii" ; format="flowed" To: r-help-request@lists.R-project.org From: "Dr. Chris Wills" <cwills@ucsd.edu> Subject: Questions about sorting and functions Cc: Bcc: X-Attachments: Dear R-Gang - Two questions for you: 1) I cannot figure out how to sort one column in an array, and

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

connecting legacy PBX to Asterisk (for the very same reason, those PBX use TE-PTMP). If others could join this thread and say if they agree or not with NT-PTMP being the 2nd most needed mode, would be great. Please, do not hesitate to comment. > > > Right now, I would not preclude the possibility that NT-PTMP support > might be added, but I could not give you a concrete time at which