search for: bluemaggottowel

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

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

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

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

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

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.

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'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

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

I'm trying to figure out which 1.6 releases have the fix I'm looking for by reading the ChangeLog for each release, whether it's in the 1.6.0, 1.6.1 branches, or an -rc release. If I look at the latest -rc releases of 1.6.0 and 1.6.1 (which are 1.6.0.11-rc2 and 1.6.1.3-rc1 respectively), will that be an exhaustive list of changes or not? The reason being I'm still waiting on the

So I have two asterisk servers, one acting as a frontend and another acting as a backend interface to a PRI. All I want is for the backend to send all calls from the DAHDI interface to the frontend. Seems like switch would do that by placing this on the backend server: switch => IAX2/frontend1/inbound_context 'frontend1' is already defined with proper username/secret combos in

Running asterisk 1.8.4.2, and occasionally we'll have a call drop and the SIP retransmit error show up on the console. I actually think the retransmit error is just a symptom of something else, possibly centered around a timing issue. I tried res_timing_dahdi and that worked for about a week, then suddenly things went haywire and calls wouldn't last for more than a few seconds before

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

Hi, We've been running g.722 in asterisk 1.6.09 for awhile now,

logins Reply-To: I've sourced the groups, but haven't had any success with solving this problem. Here are the details: I have a debian/testing system running samba 3.0.0beta2-1. It was working fine with the 2.x series, until I upgraded to 3.0. The debian server (al) is a PDC for a small NT domain (isabela) with three workstations, all running win2k pro. All but one work fine

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