search for: 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

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 ex...

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...

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", "IAX2/####@voxee/1###########|30|A(beep)") in new stack -- Cal...

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 phone rings. 1) If I answer the phone, asterisk indicates the call has been answered, says something about stopping sounds, and the console speaker cu...

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 instal...

...#39;re listening to skips quickly. I don't think -- but am not positive -- that it is a question of insufficient bandwith, as I am on a Cox 5mbps/2mbps cable line that is very reliable and pretty stable. I believe I am using uLaw both to the Asterisk server /and/ from the Asterisk server to Voxee, my outgoing SIP provider/PSTN terminator. Is this a common problem? It doesn't seem like it should be, as it is a major detriment to having enjoyable, good-quality VoIP conversations and doesn't seem like it would be the "standard" for such conversations. Perhaps I shouldn...

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 should be completed and asterisk should not be listening by then...

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 applications to determine IAX2 availability. The only way I can see...

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...

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 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 and say hello it properly carries the call except w...

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.

...and Shido6 hang out on IRC. Nufone is not a "hand > holding" VOIP provider. You are expected to have some clue. This has turned > away a number of people but as I said, they Just Work. So I've heard three recommendations for people coming from LiveVOIP: nufone, teliax, and voxee. Nufone and teliax both are at $0.02/min and voxee is at $0.011/min. No monthly fee plans are available from both. I recall hearing of troubles with Nufone support awhile back. Have those been resolved? For someone that places outbound calls only, in a fairly low volume, is there a recommendat...

...hen 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", "IAX2/teliax&IAX2/voxee") in new stack -- Hungup 'IAX2/teliax-3' -- Executing Set("SIP/501-304d", "CH=IAX2/teliax") in new stack -- Executing NoOp("SIP/501-304d", "AVAILCHAN= IAX2/teliax") in new stack -- Executing Dial("SIP/501-304d", "I...

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 does not show any events, the calls just end. I am at a loss, what can I do to debug this?...

...ound(you-dialed) exten => s,2,SayDigits(${NR}) exten => s,3,Background(if-correct-press) exten => s,4,Background(pound) exten => s,5,Background(otherwise-press) exten => s,6,Background(star) exten => _#,1,Background(pls-wait-connect-call) exten => _#,2,Dial(IAX2/${OUTGOING-NUM}@voxee/${NR},30) exten => _#,3,Background(something-terribly-wrong); exten => _#,4,Background(goodbye) exten => _#,5,Hangup exten => _*,1,Goto(testdtmf|s|1) -- Any ideas why Asterisk is NOT picking up the SIP call.... And any pointers where I've gone wrong? Thanks in advance! ________...

...r channel. I'm guessing this is for inbound and outbound channels. If you wanted to be able to have 8 concurrent channels then this could get costly. Too costly in my opinion. I mean....that seems like a LOT to me, when you can go with other providers who don't limit you to 4 channels, like Voxee, NuFone or SixTel, for around the same price. I can understand the channel restrictions for inbound calls, but not for outbound calls. VoicePulse, I know you read these lists! You should be able to provide us VoicePulse Connect users with more than 4 concurrent channels for free! -------------- ne...

...; > holding" VOIP provider. You are expected to > have some clue. This has turned > > > away a number of people but as I said, they Just > Work. > > > > So I've heard three recommendations for people > coming from LiveVOIP: > > nufone, teliax, and voxee. Nufone and teliax both > are at $0.02/min and > > voxee is at $0.011/min. No monthly fee plans are > available from both. > > > > I recall hearing of troubles with Nufone support > awhile back. Have > > those been resolved? > > > > For someone that...

...irly 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 great instructions (there weren't any) but a snappy response from the time I paid in paypal until the time it appeared on my account (2 minutes) Their rates seem good as well, I'm happy with them. There was an issue with their servers ignoring the CID in...