similar to: Console Sound: Cuts out, Comes back after restart

I've got a problem with audio output from the Asterisk console. I'd really appreciate any help. I'm simply trying to dial out to a phone on PSTN. My extensions.conf entry is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN}) exten => _1NXXNXXXXXX,2,Hangup After starting asterisk and dialing, I hear a ringback tone through the console speaker, and the PSTN

Hello, I have been trying to setup a Voxee Sip termination. If anyone has extensions.conf different than Voxee suggestion. Can you please send me a copy? Thanks! Jerry Voxee web site advises to use: [voxee] exten => _1NXXNXXXXXX,1,Dial,SIP/${EXTEN}voxee exten => _1NXXNXXXXXX,2,Hangup exten => _011.,1,Dial,SIP/${EXTEN}voxee exten => _011.,2,Hangup

I have incoming PSTN lines on an Adtran 750 channel bank. Calls are evaluated by an agi script based on callerid and forwarded to an international DID through Voxee. There is an IVR at that number that asked to user to enter a selection. When the user presses a key, my pbx puts the call on hold and tries to start music on hold. What's doing this? I have no backgrounds, no listen, the call

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

I cannot find any means of passing my own Callerid using Voxee. It always comes across as NO ID, or nothing, or unknown. I could not find anything on their website about setting your own caller id in the system either. (their web account pages). Is anyone here using their own Callerid information through Voxee? thanks

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569

I am trouble finding a configuration that works for ChanIsAvail and SIP. My two providers are Voxee and Teliax. I have these lines in a macro exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee) exten => s,n,Cut(CH=AVAILCHAN,-,1) exten => s,n,NoOp(AVAILCHAN= ${CH}) ; Dial the available Channel exten => s,n,Dial(${CH}/${ARG1},60,t) Looking at the execution, I can see what the AVAILCHAN

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

Hello all, I am trying to find out if anyone has a provider that is good with dtmf playback using a Sipura 2100? I've just dialed with voxee and the call goes through but when I press 1 my dialer does not " hear" it. My dialer is making the call using a Dialogic d/4PCI connected to the Sipura 2100 through voxee and I am calling my landline. When I pick up the landline

I searched a bit, but have come up short. Are there any libraries for creating popup forms w/ rails? These would not be displayed in a separate browser window, but rather made visible over an open page and adjacent to a clicked link -- similar to the google maps baloons, or the gmail popups. Lots of other examples out there... Thanks --------------------------------- Talk is cheap. Use

Hello all, I am using a headset and the X-lite softphone (sometimes I use IAXComm, but I'm having difficulties using OSS emulation with it) to connect via uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 megs of RAM. After getting this headset, most or all of the echo people on the other line were complaining about is now gone, according to them. However, every

I''m looking for a good way to search a few fields accross multiple asociated tables (i.e. find ''friends and family'' accross Photo.name, Photo.description, and Tags.name where Photo has_many tags). And, ideally there''s a competent query analyzer/parser. I''ve expirimented with constructing my own SQL using ... LIKE %term1% ... etc, but the

1) I'm working on a quick replacement for DISA, and I ran into the following snag: When I specify "Playtones(dial)" I can only get around 7 seconds of wait time before the dialtone stops, and the context goes to the "h" extension. Is there a way around this fixed timeout? The DigitTimeout setting doesn't seem to have any effect at all on this hangup problem. I

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

Hi everyone, I'm in the very early stages of rolling out an asterisk box at work, and one of the things I'm setting up is a trap for telemarketers >;) What I want to do is have a sipgate number in the UK here which rings for 10 seconds without calling a hard or softphone, then goes to a voicemailbox. The problem I'm having is that Playtones doesn't seem to be sending any

Anyone having problems with voxee since last few days or is it just me ? In peek hours i get LAGGED when i do a iax2 show peers or even 1000 ms latency . Most of time it is 20 ms or so but when i start sending traffic to them latency increases to 1000 ms or even LAGGED ( also shows high in peak time even when no high latency ). No problems with any other provider . Anyone else having same problem

For outbound only, I have traditionally recommended VoipJet. They just recently has a spat of issues that seem to have resolved though. I am still using them via their east coast server and it seems to work quite well again. Cost is around 1.3 cents minute I believe. Use IAX and g711 for best quality to VoipJet. Thanks, Wiley -----Original Message----- From:

I am trying to use ChanIsAvail to detect the best route for a call. I am testing by dialing an extension that is then forwarded to the DID. Normally it will be an incoming PSTN call that is forwarded. When I try it, I get put on hold for a few seconds and miss the beginning of the recorded message. Any ideas what is going on? -- Executing ChanIsAvail("SIP/501-304d",

Anyone noticed the snom 360 indications are incorrect for US zone? menu->preferences->tone scheme->usa indications.conf: [general] country=us extensions.conf: exten => 1111,1,Answer exten => 1111,n,Playtones(dial) exten => 1111,n,Wait(30) exten => 2222,1,Busy exten => 3333,1,Answer exten => 3333,n,Playtones(busy) exten => 3333,n,Wait(30) hit speakerphone on the

Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten => 1003,1,Wait(1) exten => 1003,n,Playtones(!1050/1000) exten => 1003,n,Wait(1) exten => 1003,n,StopPlaytones exten => 1003,n,Echo exten => 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones