similar to: AutoDialing with VOP USING SIPURA 2100'S

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

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

One of my PBXs drops calls after 7 to 10 minutes. I cannot see any reason for this. I upgraded to asterisk 1.2.7.1 last night, still no improvement. Calls are IAX2 to either teliax or voxee, doesn't seem to matter which. Codec is G729. Connecting over ADSL. Load is only onw or two calls, server is P4 2.4 GHz. Monitoring the ADLS does not show any significant packet loss. Watching the CLI

Hi People, I'm writing to get some advice on where to start when learning asterisk? I was going to begin learning with AAH but I wanted to find out if there is a certain build to avoid or if there is a Gui/front end that is better then another. I have been working with dialogic cards for the past 5 years and with auto dialers but I want to get into providing voip service, support and

Hello, Anyone here have experience with Audiocodes MediaPack MP-108 Gateways? I would be willing to pay someone for advice and support with configuring my gateways for a telemarketing project I am starting. My experience is somewhat limited but all I want to do is make outbound calls just like I would on normal pots lines. (That's the best way to explain it) I do not need any special

I am looking to see if anyone has any info on auto dialer software that connects directly to a voip provider without using any third party boards or digium cards? I've been building dialers for the past 5 years and I want to get out of using add on cards and just make calls from the software directly using voip. The software would need reporting features, answering machine detection, hangup

<http://www.vistaprint.com/vp/gateway.aspx?S=5176697856> Anyone on the list good with Linksys PAP2NA configuration, I am looking to take my ata's and emulate the operation of a pots phone line as close as I can get. One thing I need to change is the fast busy tone I get when someone hangs up on the call. We are also looking for a Voip/ Asterisk Consultant to set up hardware for a

Hi, All of you people, I just want help on issue regarding auto dialing in asterisk. I have implemented asterisk server using TDM400P with four FXO Modules, as well as I am using analog phones. That's works fine, can any one of you will guide me, how can I implement basic auto dialing functionality in which I will be storing list of phones number either in GUI interface or will feeding in

Hi, I am just about to move out from my parents home and think about how I will phone from now on. In Germany there is a provider (QSC) who offers DSL (1024 down/256 up) with fastpath without volume or time limits. Does anybody know a comercial (or even semi-professional) provider who lets me dial out through H323 (or another protocol) and also offers an number where I can be called from

I'm having a problem with sound output to the console. My basic dial plan is as follows: exten => _1NXXNXXXXXX,1,Dial(IAX2/####@voxee/${EXTEN},30,A(beep)) exten => _1NXXNXXXXXX,2,Playtones(info) exten => _1NXXNXXXXXX,3,Hangup I get the following output in the console: ___*CLI> dial 1#######@voxee -- Executing Dial("ALSA/default",

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

<For someone that places outbound calls only, in a fairly low volume, is there a recommendation for which one would be <best for me? <I have had continual audio trouble with LiveVOIP, though other services <(FWD) work fine, so I'd want something that has good audio quality. I will toss in my $0.02 and say that I have had good luck with Voxee, simple setup, good quality, not so

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

A few recent revisions will likely break your builds, since svn is having trouble with the deletion and then addition of $SHS/vop. If you have errors when updating, try removing $SHS/vop and then update again. Andrew

On Fri, Feb 22, 2019 at 04:30:52PM +0100, Vincent Whitchurch wrote: > Add a loopback driver to allow testing and evaluation of the VOP > framework without special hardware. The host and the guest will run > under the same kernel. > > Signed-off-by: Vincent Whitchurch <vincent.whitchurch at axis.com> > --- > drivers/misc/mic/Kconfig | 10 + >

tree: https://git.kernel.org/pub/scm/linux/kernel/git/mst/vhost.git vhost head: c4ea43c2c1779f5dde3ff5645b830c90f75ccc15 commit: efd7eb77f631e1ed3533db7725df157a379c78ef [11/15] VSOCK: Add Makefile and Kconfig config: x86_64-randconfig-x005-201631 (attached as .config) compiler: gcc-6 (Debian 6.1.1-9) 6.1.1 20160705 reproduce: git checkout efd7eb77f631e1ed3533db7725df157a379c78ef