similar to: Loud echo with answer before dial

Because I'm tired of seeing Debian users look like idiots for using such horribly out of date versions just because they run "stable", I've set up a page with .deb packages of my 1.1.x rebuilds for use under lenny. http://www.rollernet.us/opensource/ Standard disclaimer applies: use at your own risk, no warranty, no support, if it eats your first born or erases your computer

I've come across an odd dialing problem with my * setup. After * has been running for a while, if I try to dial out on any of my zap channels, (both are X100P cards) it picks up the line but never sends the DTMF. Has anyone heard of or seen this problem before? Right now I'm looking at 19 hours of uptime on * itself. If I restart it, everything works fine for a little while, then the

Does anyone know how (if possible) to do three way calling on the UIP-200? There doesn't seem to be much info about this phone, but all the feature lists I've read says it can do conference calls. I can't seem to do it, though. Any help would be appreciated. -- Seth "et lux in tenebris lucet" Mattinen sethm@rollernet.us

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

I've been trying to get fax reception to work using an SPA-2000 to ring the fax machine or modem that's taking fax calls. I was curious if anyone else had tried something similar, and if so, had any luck getting it to work reliably. I've been able to get it to work, but it isn't reliable. (Pages/lines of black dots result more frequently than not.) The incoming lines are FXO

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

On Jun 20, 2004, at 9:16 PM, asterisk-users-request@lists.digium.com wrote: > I am new to Asterisk, but though that I would need calls being > answered in > different contexts. > > How can I direct one line to a given context and the other one to > another, > or is there a better way??? > I use this at the end of my zapata.conf file to do exactly what you're

Perhaps I was a little too hasty in my conclusions of dysfunctional fax on the SPA-2000. It turns out I have a one way audio problem on line one of my SPA-2000. I have all the correct settings according to the comments in the wiki, but the problem persists. However, if I do a hook flash out of and back in to the call that isn't transmitting audio, it works fine. My sip.conf entry for the

I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP -->Asterisk server-->Natted firewall-->extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except

T100P with an Adit600 channel bank. 16FXS, 8FXO. Been doing this with the past month or so's worth of CVS HEAD, probably longer. I have a weird problem when I bridge calls... not always but often enough to be nasty. Call comes in FXO, * calls my home via IAX2... times out so picks up another FXO port and dials my cell. 7 times out of 10 it bridges just fine, but about 30% of the time

I have the following dialplan (segment) that isn't working as I expected it to: exten => s,n,Dial(Zap/1&SIP/202&SIP/203,18) exten => s,n,Dial(Zap/1&SIP/201&SIP/202&SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

I've got this in extensions.conf: [macro-stdexten] exten => s,1,Dial(${ARG2},30,p) exten => 6015555555,1,Macro(stdexten,200,SIP/200&SIP/201&SIP/203&SIP/${VOICEPULSE_GATEWAY_OUT_A}/+15045555555) Where the real numbers have been replaced with 5555555. What I'm trying to do is ring my cell phone in addition to the local extensions. Funny thing is the cell phone rings

Crew, I've run into something a little odd with the vorbis encoder and would like some input. I use SpacialAudio's SAM3 Broadcaster and am having problems with the ogg encoder when I create a mono output stream. It seems as if the signal level of the left and right sides are being combined before the encoding process, leaving the sound muddy and overdriven. As such, if I reduce

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took

I'm running FreeBSD 4.2. I downloaded and installed Icecast 1.3.10, Liveice, mpg123, and Lame. I know the Icecast streamer is working because I can feed it from WinAMP on my PC and listen to the stream somewhere else and it sounds fine. I know liveice is working too because Icecast shows that it connects and is sending audio to it. However when I listen to the stream on my PC with WinAMP

Anyone experience ring oddities with extensions.conf rollovers? Let me summarize... One of my extensions.conf file is built to ring during the day, ring/go to voicemail after a certain time: [main-aa] exten => s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1) exten => s,2,GotoIfTime(*|sat-sun|*|*|*?main-night-aa,s,1) exten => s,3,Dial(SIP/201,25,tr) exten =>

Hi Jean-Marc, after some problems with getting svn to work here I finally made it. Problem is, you write that I cannot use libspeex and libspeexdsp at the same time now -- because I use a "live" system (mic-in -> speex_enc -> speex_dec -> headphone out) and I can run the AD1836 audio codec on 48 kHz only, I cannot use my program now (because I use speex resampling...) So I

I've used a Sound Blaster AWE32 / Liveice combo before. Sounds like your record volume is set WAY to high or more likely you are recording from the wrong input. Make sure that it's the _line_in_ that's selected for recording in you mixer app. Also If i remember you have to use HALF_DUPLEX mode (in liveice.cfg) at least if you're using MIXER mode... I'm not sure about in