similar to: Voicepulse

Hi all, I am new to asterisk and am looking for a voip provider that supports asterisk. I am aware that their are several vendors to choose from. Any opinions on the best one? thanks Burak Balasaygun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060128/0a97a302/attachment.htm

Hi I have the following configuration at home one ZAPTEL interface connecting to an FXO card and two SIP UAs connecting to asterisk locally. I have configured extensions.conf such that dialing 9 on the SIP phones allows me to dial an outbound number via the FXO interface . Works fine. What's not working is that pressing 9 should causes either GS BT-100 phone to reacquire a dialtone

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

Hello- I'm obviously doing something wrong here in trying to get an inbound DID to work with voicepulse. I have an outbound context set-up for those calls in iax.conf, and the appropriate register in- statement. within extensions.conf I am doing something like this: exten => 212xxxxxxx,1,Dial(SIP/admin,t) (where admin is the phone i am looking to forward to from sip.conf). i'm

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

I'm a bit displeased at the way this happened. I received an email from VoicePulse. Here's some excerpts: ------------------ >We're sending you this important update so you can take advantage of improvements we've >been making to your VoicePulse Connect! service. >We've been working hard on improving the audio quality and reliability of your Connect! >service,

Sorry, I am newbie. Sample codes are from https://github.com/oneman/libflac/tree/master/examples/cpp. I used FLAC__stream_decoder_process_single function but it still gives exception. Maybe I could not control read callback, you're right. I will check it and write result in this thread. Thanks for help. 2013/7/1 Martijn van Beurden <mvanb1 at gmail.com> > I'll top-post this

Martijn, I don't use any metadata when encoding and decoding. When I call *FLAC__StreamDecoderStateString[FLAC__stream_decoder_get_state(m_decoder)] * * * it returns FLAC__STREAM_DECODER_SEARCH_FOR_METADATA enum value. Is it an error ? 2013/7/2 Burak Or?un ?zkablan <borcunozkablan at gmail.com> > Hi again, > > I can not solve problem. I want to mention my source code, so

I am trying to obtain adjusted means and standard errors for a three way ANOVA I have three effects, two continuous; fire frequency and annual precipitation, and one categorical; soil type in an unbalanced design. I am testing the effect of annual precipition (AP), soil type (ST), and fire frequency (FF) on stem count (SCt) My data table looks as such: ST FF AP SCt 3 Coy

Hello, I was wondering if there is a way to ignore the error message you get when some of the data means you compare are constant in some lines of your data frame. I'd like to go ahead with t.test and get the calculated p-values anyway in such a case. Thanks -burak

I am using Asterisk CVS from last week and have been using Realtime SIP for a couple weeks now without any problems. Yesterday I decided to turn on Realtime IAX but I am having problems dialing to my long distance providers like Voicepulse, Sixtel or Nufone. I get the following: -- Executing Dial("SIP/2001-3761", "IAX2/password@voicepulse/19566680301") in new stack

I have two questions regarding number portability... 1) If I port a DID over to Voicepulse, can I then move that DID elsewhere somewhere down the road. Or does voicepulse now OWN that DID? 2) Can I take a DID assigned by Voicepulse and transfer it to someone else? If not, why? -jwb

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

Hi, I am developing an audio network system and using boost, OpenAL and FLAC library in C/C++. I can stream raw audio data over network but I want to encode audio before streaming in current PC and decode after streaming in other PC because of bandwidth limit. I run your sample codes, encode.c and decode.c, about file encode / decode. Then, I run streaming encode / decode with two different

I'm looking to set up multiple inbound Voicepulse Connect lines and have Asterisk route them direct to different IVR or Voicemail based on the inbound number that is called. Unfortunately, I just can't see how one would go about identifying the number that is being called. Has anyone been able to do something like this with Voicepulse? I appreciate any assistance. Phil

Has anyone been able to get * working with VoicePulse OpenAccess (SIP not IAX). I have found a ton of information about VoicePulse Connect but very little on the proper * settings for OpenAccess. Tried contacting VP with no response. If anyone has this working, can they share their extensions.conf and sip.conf files? Better yet, if it could be posted on the Wiki. Keith

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

The majority of sip to pstn gateway providers (vonage, voicepulse, and others) appear to be setup for a one line only type of set up. Their web sites seem to be heavily geared for these one line setups. Anyone willing to comment on what type of pricing plans these providers offer when using iax2 trunking or other methods with asterisk to send multiple (and possibly simultaneous) calls through

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Dear Sirs, Anybody ever tried running SIP up against Voicepulse? On their http://connect.voicepulse.com they claim they support both SIP and IAX, but I can't seem to get SIP running. I have as mentioned before on this list - huge problems getting any timing devices running on some of my machines, so IAX is not really an option right now. If I try I get a "Service Unavailable"