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When I call in from an outside POTS line to a Zap channel, and the call ends, it seems like the hangups are very "sloppy." I see Asterisk give the hangup command, but on my phone there's lots of clicks and the line acts like it's staying open for several seconds, then I hear a phone ringing sound followed by "If you'd like to make a call, please hang up and try

Most of the time, when someone calls in from the outside on a POTS line, and possibly over IAX as well, they don't hear any ringing sound while the internal SIP phones ring. If you call from an inside SIP phone, even forcing it into the incoming context, you hear the ringing. The outside calls can answer and talk fine; just no ring indication. Is there a setting that controls this?

I see that when an Asterisk connects to another one via IAX, it seems to use port 4569 for the first one. But if it has multiple IAX connections the additional ports seem to be chosen at random. Is there anyway to predict, or specify which ports or range of ports to use, for the sake of setting up a firewall? Thanks.

I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines?

I've been trying to search the archives for older messages, but the archive at: http://www.mail-archive.com/asterisk-users@lists.digium.com/maillist.html only seems to go back a few days. Is there another archive somewhere that goes back farther?

...onnected to asterisk TDM cards. Thanks a lot. Michael ======================================== Hi, Oops, my bad. Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise. Message: 14 Date: Wed, 03 Dec 2003 17:43:16 -0500 From: Matt Lawson <matt@1control.com> To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo cancel in MeetMe? Reply-To: asterisk-users@lists.digium.com I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into...

Hi everyone, I know someone makes a product that's a POTS phone to SIP converter, where you just plug your POTS phone in one side and the network cable in the other. Has anyone successfully used any of these with Asterisk, and if so how expensive were they? I ask partly out of frustration with the FXS cards but mostly because it would make installation MUCH easier for what we're

Hello All: Has anyone configured a meetme conference to use video? I have successfully used video phones to talk through *, but I cannot seem to get video when those phones dial into a meetme conference. Is there something else that I need to be doing other than set the "v" flag on my extension for the meetme app? Thanks, Tim

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

> But this is not to say _you_ can't built a reliable VOIP based > system. Get _two_ providers and set up your dial plan in > extensions.conf to "fail over" if one service fails to > connect to dial via the next one and finally if both fail > use pstn. your users will see a system the "just works". Now there's an idea. I'm playing with this now,

I have discovered something quirky in our Asterisk. If I call in to a Zap channel (from an outside POTS line), then transfer the call around several times, I get the above error, after which it will hangup. I believe Asterisk may issue a SIP CANCEL to the extension it was starting to dial. Now when I say 'transferred around several times,' our routing is pretty compex and uses the

We have some regular POTS phones connected to our incoming line as well as the machine that runs Asterisk. Sometimes during an outgoing call from the POTS phone, the Asterisk will pick up also, and play its menu. The FXO card is set to fxs_ks signalling; I'm told this might be the culprit but I really don't understand about the signalling types and what the ramifications of

Hi everyone, I'd like to set up the RSA keys for the IAX registration, but have a couple of Q's. I have the manual and can follow the instructions, but I want to understand the limitations. First, understand there will be a central Asterisk (which has the private key?) and several remote Asterisks (which are as automated as possible, and each have the same public key?). We

Is there a way to determine which channels belong to fxo vs. fxs devices? I need to write an auto-configuration program that can match up channel numbers to types. I have to assume there's an unknown ordering of fxo and fxs cards. Suggestions? TIA

I have one of the Digium 4-port FXS cards, although it only has 1 of the 4 modules installed, making it a 1-port FXS card I suppose. I plug it in, and plug the power connector into it, but none of the 4 LED's come on. Are they supposed to? doing an 'lspci' I believe I see it listed as a "Tiger Jet" product. But when I do a ztcfg or cat /proc/zaptel/*, it doesn't

How can I loop through something x number of times in the dialplan? i.e. if I get an invalid extension I want to re-play the menu, but not forever. Maybe 3 tries or something. I'm pretty sure that I've seen it before, where you can increment a variable and do "Gotos" based on it. But I've searched the Asterisk handbook, searched the user archives, and Googled for it,

Hi all, I'm wanting to take an existing call, and transfer both sides of it into a meetme room (yes I know the phones have a conference ability built-in but humor me). What seems to happen is I can transfer one half of it fine, but as soon as I do that the other half hangs up. Do I have to park it briefly? If so, what does the call ID become once it's parked, so that I can

Hello, I'm trying to compile a brand new CVS Asterisk and running into trouble with include files. I have an older version of Asterisk that I can compile (2-3 months old) that I can compile fine, but the new one gives me this: make[1]: Leaving directory `/home/matt/asterisk_update/stdtime' if [ -d CVS ] && ! [ -f .version ]; then echo CVS-10/01/03-13:06:31 > .version; fi

With a recent update to Asterisk CVS, and versions 2.02r and t of the Snom 200 firmware, I'm getting the Snom phones stuck reporting "Busy.": -- Got SIP response 486 "Busy Here" back from 10.12.34.248 -- SIP/3064-b07d is busy They're on-hook, not doing anything. They are registered fine. I can pick them up and call out over Zap or IAX fine. I just

In an attempt to reduce bandwidth usage, I tried forcing my Asterisk to use Speex. I did a "disallow=all" then an "allow=speex". The crazy thing is, it didn't reduce the bandwidth usage at all! I can do an IAX2 show channels and it shows the call being in format 512 (Speex, right)? Then I switch iax.conf to only allow ulaw. I retry the call. IAX2 shows the call is