similar to: Skipping problems on outgoing calls (using uLaw with an internal * server through Voxee)

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

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?

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 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

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

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

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

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

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, I have installed the nlme package, but every time I try to use the function "summary.lme," I get a message that the function cannot be found. I've tried to install the package several times and have re-started R several times, but to no avail. I have also loaded the lme4 package, in case that makes a difference. I'm using R on a brand new Mac. Thank you!

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

Hi, guys. May be someone could give me advise? I am trying to find good VOIP provider ONLY for OUTGOING calls with low per channel cost and cheap rates on Eastern Europe, Turky and xUSSR. Should support g729 or g723 codecs, SIP or IAX connectivity. -- ========================================================================= = Best regards, Nikolay Pavlov.

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

?I need to install Asterisk on a MAC, native, no virtualization. Has anybody done this? Are there documents on the Internet? I googled it and all web sites that claimed to help installing Asterisk on a MAC have disappeared. Is it possible at all? Digium should actually have a MAC app in the Apple store with a PBX. It should be a paid app. I would buy it right away. ? -------------- next part

Does anyone know what happens if you try to have 5 concurrent outgoing channels with VoicePulse Connect? Does it give you an error message or a reorder or something? I'm worried about using them as my primary carrier if this is the case. I noticed that they supposedly only allow 4 channels for free and then you have to pay $20 a month extra per channel. I'm guessing this is for inbound

Hi, I am running several Asterisk boxes with realtime around the world. Does anyone have a recommendation for a "light" db that would work with Asterisk that would also allow replication between all sites (so if I add an entry to one box it will work with the rest)? TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have a user who would like to stream their favorite radio station from iHeart radio for their music on hold. It this TECHNICALLY possible? If so, any pointers would be appreciated. Is this LEGAL in the US? Thanks in advance, Mike. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com> wrote: > Hi Dovid, > >