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The scenario we have is fax send/recieve software that ONLY talks T38 and an asterisk box. We have ITSP providers that do NOT talk T38 but G711 only. Does asterisk have the capability to take the T38 call from an ATA or T38 software then bridge/transcode it and do G711 out to the PSTN providers? If not is there another product PAID or FREE software or hardware that can do this easily and

Trying to compile asterisk (again) from scratch. I seem to be still experiencing the effects fro Jan 25 where I get no sip to sip audio. I have tried upgrading to 1.2.3 which has made no change in the problem. I am starting over and now trying to compile/install /trunk zaptel libpri asterisk following the instructions to grab the source trees: # svn checkout

Sorry for being a bit of a newbie here but I find the docs or README for downloading the G.729 codec from Digium are not as detailed as I would like or just don't really break down the different versions to a point that I am clear on which one to grab. The choices for 32bit are: drwxr-xr-x 3 0 0 4096 Dec 05 00:21 athlon-xp drwxr-xr-x 3 0 0 4096 Dec

I would simply like to dial an extension and get an individual Live MP3 stream but am unsure of how to do this. I'd like it to be different from my music on hold (not the same source) This trick works for music on hold: in musiconhold.conf ;default => mp3:/var/lib/asterisk/live,http://sourceofstream.com:8001/ I still wish to use local files for music on hold but want to dial an

I keep running into the dead end that it can't find config.h in the source tree. It looks like newer kernels don't use it anymore. Someone ran into this awhile back when compiling 1.2 and it looks as though the issue was never resolved. Any ideas? Last time I tried this, I was on fedora core 5 64bit and all went well. It's not working on this newer setup Any ideas on what I can do

been testing with a rather simple setup. The mission is to actually get a reinvite to work on the lan. I am trying with two sipura phones G.711 codec forced on both both on the lan no nat no fancy options suchs as tT or H No matter what we do asterisk hangs on to the media path, how in the world do I get a reinvite to work where the media path is actually handled by the two phones on the lan?

Sorry this is a bit of a newbie question, I've been at this for a few months and still have not quite figured this one out. I've been able to setup one itsp (incoming calls) (sip account) with a register line like this: register => nnnnnnn:ppppp@sip.provider.net -or- register => nnnnnnn:ppppp@sip.provider.net/nnn to come directly into an extension in the dialplan It seems that

This is a very newb. question. Been using asterisk very happily now for several months and am considering getting some of those really 'cool' multi-button phones with LEDs and buttons. It's unclear to me if it is a straightforward task to actually setup a multiline 'presence' on the phones where the LED's light up when someone picks up a 'line' or is using a

What we would like to see happen or emulated is that if someone makes a call via our SIP provider to a PSTN number that is actually busy that we get an actual BUSY tone at the telephone. In our test case this is a PAP2-NA SIP device It would appear that when we call the far end (PSTN phone number) that is busy we do not get any busy indication at the user end (originating telephone on our

Still looking for some direction with this subject: I think the term is called multi-line appearance.... Is this something that is directly supported in Asterisk? I can't seem to find any information on it or how to actually use it.... This is where you have several sipura-841 SIP phones for example... if someone pickes up 'line1' I'd like the light to come on on ALL phones to

I'm having difficulty understanding this 'hint' feature of asterisk. My limited understanding is that it is somehow needed for 'informing' some kinds of phones that can do shared line appearance to show the state of the channel/user... Is this true? the wiki has this: http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence I'm still having much difficulty

On the same P2 450Mhz box..... I have tried both UHCI usb on a 2.4 kernel and enhanced RTC on a 2.6 kernel. Have not tried UHCI USB on a 2.6 kernel as of yet. Both seem to work GREAT. I have read in many places to be sure to use a digium card as a time source and not to reply on the cheap solutions. However I have regular meetme sessions of 5 and 6 people at the same time that frequently go on

This was just a recent personal experience.... Maybe I missed a thread on this: We recently installed asterisk (CVS-HEAD) on a debian system using 2.6 kernel and the enhanced RTC for all timing. Also a custom compiled kernel for the CPU on the box (P4). We had a strange thing happen in that with Debian's MPG123 package: Sound files played in asterisk/mpg123 were heard at literally 1/10th

Hello, I'd like to know if it is possible to get * to listen and respond on more than just one single udp port. I've run into several situations where I'd like IAX to work on an alternate port as well as be able to work on the standard port. I'm wondering if there is a way to do this? Thanks!! Steve

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

I've recently installed the latest Debian Linux for powerpc onto and old iMac (version A) the original iMac with a 233Mhz G3 processor and 160MB of sdram. The debian install went smooth and so the the apt-get install of Asterisk 1.4.21 It appears to have no functioning zaptel or ztdummy module. Is because of hardware? or is it because whoever built the package didn't include a full

I'm looking for some help in how to keep asterisk from doing this. If we loose Internet or routing to our upstream provider even for only a few short minutes asterisk quickly gives up & never tries again. I have to do a manual reload to get it to register with my sip provider(s) again before incoming calls are accepted. This is really bad as it causes us to loose the ability to get

I am using asterisk (2 week old CVS) am for the first time have been starting to experiment with busy and congestion. At this point I am only using sip endpoints PAP2-NA devices. All testing of this is being done on a local network. my test extension looks like this: exten => 7777,1,Answer exten => 7777,2,busy(35) exten => 7777,3,Hangup Or like this: exten => 7777,1,Answer

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

I have a small pile of them from customers who were too lazy to send them back after switching to our local voice service... Is there any hope of ever using these things with Asterisk? Vonage does not want them back and they won't unlock them either. A terrible shame! Should I just toss them? Steve