similar to: switch statement in extensions.conf

Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that

Hi, I have two Asterisk servers and I forward calls from one to the other. How do I reload extensions included in a switch statement in extensions.con? I have tried "extensions reload", "reload" and "restart now", and it's only "restart now" that works. Is this how it is supposed to work or can it be a misconfiguration? Regards, Mikael Magnusson

On the extensions.conf explanation page is a mention of the "switch" statement and it refers one to the "connecting two * servers" page. The only mention of the switch statement there is brief and in the example. However, the example seems to have some errors in it. It shows a sample of what's in the extensions.conf file, but it clearly has sections which would be in the

Hi all I'm trying to understand what the switch => statement in the extension plan exactly does. Unfortunately it seams to be very badly documented. There's allmos noting about it in o'Reilly's Asterisk Book. Nothing in the shipped /doc, nothing on voip-info.org. Even with google I found ony very few infos. So can somebody explain? I got two Asterisk Servers running

Hi all, i need to know how to use the switch statement in extensions.conf to throw calls on a secondary asterisk server. I have been trying all day to be able to make it work but its not working. It displays the following error: pbx_find_extension: No such switch 'master:secret at 192.168.0.7' I have read the help material on voip-info for connecting 2 asterisk servers and followed the

I can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten => conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf => 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the

We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI card in it connecting to the PTSN. The frontend handles 99% of dialplan logic and just hands off anything outgoing to the backend via IAX2, which dials out on one of the open channels. Lately we've been getting a disconnected calls. Keeping the consoles running it doesn't seem to be

Hi, I've always used dahdi-genconf to just create the dahdi-channels.conf and since our PRI is fairly simple (just dump all the channels into one group) it works with dialing with dahdi/g1/(number). I'm trying to understand the file though for my own reference. It seems the file looks like this: group=0,11 context=from-pstn switchtype = national signalling = pri_cpe channel => 1-23

Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of

https://bugzilla.mindrot.org/show_bug.cgi?id=1039 Iain Morgan <imorgan at nas.nasa.gov> changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |imorgan at nas.nasa.gov --- Comment #13 from Iain Morgan <imorgan at nas.nasa.gov> --- My apologies for

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

Hi, I'm getting the following error from an asterisk 1.6.0.9 installation: [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler This repeats a few times, then asterisk crashes. I can't seem to locate any info on this error at all. I'm using

Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I changed something in features.conf and suddenly any time I hit transfer (*2), I can only enter one digit before asterisk immediately tries to dial that extension.

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there

Hello, If I put two spans' worth of channels, say 1-23 from span 1 and 25-47 in span 2, in one group, but only span 2 was showing OK and the other was down / showing a RED alarm, would asterisk automatically skip over trying to use channels 1-23 when doing outbound calls? e.g., dial(dahdi/g1/(number) would just jump to channel 25? Testing seems to bear this out, but I'm not positive

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer Can someone clarify what "early media" is? I noticed that NOT answering a call before dumping them into a queue that has music on hold will not set up a leg to push music back over the calling SIP channel. Tossing an Answer command into the dialplan just before moving to the queue alleviates this (in either situation the

Can anyone provide more information on switch or point me to where I can find more about it? The only I've been able to find on the wiki is: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers and towards the bottom of (section Forwarding to another Asterisk): http://www.voip-info.org/wiki-Asterisk+config+extensions.conf Some of the questions I have are: 1) If I have an

I've recently decided to spend idle cycles while waiting for various Astlinux platform builds to complete on making the contents of asterisk/config a little more complete, a little more useful, a little more real-world... I started looking at the possibility of taking JTodd's ISN Freenum cookbook example and updating it: http://freenum.org/cookbook/#ASTERISK_CONFIG but ran into some

Hi, I'm setting up multiple asterisk servers and trying to do the classic DIAL(IAX2/asterisk1/${EXTEN}&IAX2/asterisk2/${EXTEN}&IAX2/asterisk3/${EXTEN},15) After googling a bit, I fell on a discussion about putting this in a variable so that adding additionnal servers would be easy. I can't seem to find the link anymore, but it went something like this: extensions.conf:

Hi all, I'm a bit confused about how to use the switch statement. I've got an IAX2 link between 2 servers (SA & SB). I have use the switch statement to include extensions from SA onto SB which is happening perfectly. I've read that I can't use the switch statement the other way (ie SB->SA) at the same time. What is the correct way for both servers to know about each others