similar to: IAX calls dropping after minutes

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

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

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 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",

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:

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

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

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

One might also conclude that during the outage the support people were focusing on getting the system back up and were not near phones. At least that is what I would bet on. Just a thought considering how most of the smaller ITSPs seem to work. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

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?

Anybody using Cisco Call Manager and connecting to any SIP termination service like voipjet, voxee, etc? Please msg me offlist. AK

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

Quoting Richard Pennington <rich at pennware.com>: > I've just started looking into code generation and have a newbie > question: Is there enough information in the .td files to make a tool to > automatically generate an assembler from them? Is a project like that > in the works? > > -Rich Hi your question is reasonable, but it is probably out of scope for LLVM.

Hello everyone. I can place outgoing calls no problem with my IP500 (using teliax as our provider). Thing is, when a call comes in, 90% of the time when I pick up the handset it drops the call immediately. I turned on SIP debug, and have listed my extension config from sip.conf. Any help is greatly appreciated.... sooo close.... TIA! -Ron [3004] type=friend username=3004 password=XXX

Hi, Nikolaos Kavvadias wrote: > Quoting Richard Pennington <rich at pennware.com>: > > I've just started looking into code generation and have a newbie > > question: Is there enough information in the .td files to make a > > tool to automatically generate an assembler from them? Is a project > > like that in the works? > > your question is reasonable,

Hi I am using Rserve for R2.3.1. every time after I load Biobase library, a new Graphics window frame pops up. Could any onw know how can avoid it. Best Saeede class testReserve { public static void main(String[] args) { RServeConnection rsCon = null; Rconnection c = null; Process proc = null; try { Runtime rt = Runtime.getRuntime(); proc

Good evening all, hope everyone in the US had a nice Thanksgiving! On one of our internal servers, I decided to make the leap from 1.4.2x to 1.6.2.0-rc6 so I could start learning about the changes and new features that have been implemented. I upgraded all the configs, removed all the deprecated stuff, etc -- well went well. However, I noticed after the upgrade, when dialing into an

We are unable to call certain 800 numbers through Teliax but I thought I would post this here and see if anyone else had the same problem with either Teliax or other carriers. The 800 numbers causing problems pick-up the call right away and are in the US - American Airlines (8004337300) and Staples (800-378-2753) - we can call many other 800 numbers just fine. Our asterisk setup has a 4-port

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt