similar to: Terrible inbound call quality vs. outbound

I've been having random problems when I make long distance calls using either VoicePulse or Nufone. Sometimes the calls go through clear, and other calls (or even just part of a call) the person on the other end just hears garbled voice, or really broken up voice. Sometimes it lasts for only a few seconds, but other times it goes on for a few minutes until I give up on the call. At

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

My goal is to have only 1 primary phone number that can seamlessly "pool" multiple VoicePulse accounts. Let's say I have 3 accounts with VoicePulse Connect 212-555-1000 (primary) 212-555-1001 212-555-1002 When I receive inbound calls on 212-555-1000, I want to "forward" or "roll over" the connection to 212-555-1001 and 212-555-1002 so that the 212-555-1000

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

Hi there, Since last 2 days iconnect's incoming is not working. Is it the same with everybody? For the past 5 months I've been using this service perfectly in two boxes and suddenly it stopped functioning. I'm able to call out, the version is 0.9.1. Any help is appreciated Thanks, Raj --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB

What I am trying to do is in the 3rd option dial my cell# thru voicepulse I just can't figure how to construct the line [inevans] exten => s,1,setcallerid(${CALLERID}) exten => s,2,Dial(MGCP/aaln/1@Egraph-1,10,tr) exten => s,3,Dial(iax2/passwod@voicepulse.com/ Where do I put the # to dial 18708573287 thanks James Schenck Egraph Design Inc. Arkansas Online Internet Services (870)

I'm trying to get ChansSpy to work. It works, in the pass/fail sense, but it is difficult to understand the various speakers. I can hear users on our end just fine, but the other end sounds like their going through a vocoder, if I can understand them at all. Otherwise it is just garbled. We are using the following setup: all of our phones are SIP phones; for our outgoing calls we make use of a

I am trying to do the following: 1. Call comes in to my * box over IAX (VP Connect DID) 2. Check to see if call should be forwarded to my cell 3. Forward the call to my cell phone and take * out of the media path. I am able to do all of the above except * is not able to natively bridge the call. I am using sixtel and for the call forward portion, but the calls don't connect before sixtel

Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, "In what format do

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and

Has anyone set up Group/Broadcast Voicemail for 50 or more mailboxes?

Is anyone else having really bad problems with Voicepulse over the past few days - terrible call quality, dropped calls etc.? Or is it a problem our end? Arrgggh. Makes you want to go back to POTS :-( Simon -- Simon J. Coles President & COO, Amphora Research Systems http://www.amphora-research.com/ EMail: simonc@amphora-research.com Phone: (513) 697-4764

I'm hoping someone can help me with a problem I've been having for a while now. I've googled and wiki'd to no avail. Whenever I place an outbound call from * to a PSTN through a SIP or IAX provider (e.g. Voicepulse or Broadvoice), the first 1/2 to 2 seconds of the remote call are clipped (muted). For example, if I call a remote voicemail system that usually answers with

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hello, I am testing speex 1.1.6's echo canceller. I am using testecho.c, with a few modifications to get it to run on Windows. My problem is that I am unable to get the echo cancellation to work correctly. I am working on an audio conferencing software, and one issue we have is sometimes the microphone picks up what is being played through the headset, resulting in an echo of the other

Just wondering whether we have a resolution to iconnect incoming problem, which started few days ago. Cheers SW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040808/ecc99c4a/attachment.htm

Hi All... I have a need to phone a large number of people and collect information from them. I know Asterisk has a nice IVR system, but can it be used to initiate a call to people listed in a database or text file? Don't worry, this is not an annoying marketing thing. Thanks...

Hello: I am new to the list. I am trying to set up asterisk with voicepulse. I have a voicepulse username + password, and SIP DID. When I login to voicepulse, I have this under my devices tab: Devices *Login:* Sysxxxxxxx *Password:* xxxxxxxxxx *Context:* VPWS *Connects to:* gw5.voicepulse.com My question is: Do I need a 2.4.x kernel? Currently I am running Debian/stable stock 2.2.x ? Has

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

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