similar to: Missing first second of voice on outgoing SIP/IAX calls

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 know that the wiki has an extensive list of European VoIP providers out there....but there's so many that it's kind of hard to sort through. So I was wondering if anyone could recommend some reliable SIP/IAX termination providers in Europe? Something like VoicePulse Connect, NuFone, Vitelity, or Junction Networks based out of Europe. I really don't trust a US VoIP company for

Hello, I hate to ask here, but.. Does anyone know of an IAX/SIP DID provider in Vancouver, British Columbia? I'm looking for a voicepulse isk service, one DID with standard calling features and some sort of long distance package. I've looked around on voip-info.org's list of VoIP providers but so far I haven't found one that offers a 604/778 number. Thanks Matthew

I have been looking at moving from SIP-based DID (Illinois) providers to one that uses the IAX protocol for DIDs. After a search, I've come up with the following: http://connect.voicepulse.com -- $8/month, many rate-centers http://www.iax.cc -- $1.50/month + 0.014/min, many rate-centers Can that be all that there is? I like the pricing plan at iax.cc, because it would allow me to set up

Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back.... But how to properly handle this for iax, sip calls.... I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive

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

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

I get these on a consistant basis for most of the providers I have configured. Some less than others. I even get it from my phone at home to my * box at our data center. What I'm confused about is why it always shows the ping times at right around 2000 ms. That just can't be right. It's always right at 2000 ms. Never less or more by more than 100 or so. May 12 17:42:23

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

I am terribly sorry to bother the list with such generic and bizarre problems, but I have been racking my brain with these for the last week working on it for at least 60 hours. If anyone can even point me in the right direction I would be eternally grateful. So without further adu here are my woes: I have * (2004-04-09 CVS) running on a P4 1.6Ghz CPU, 512MB RAM, Debian "Sarge", and

Hello- I am in the process of adding a new provider to my asterisk box (both for outbound termination as well as inbound DID). They are going to be delivering and receiving traffic via SIP only. Now, in IAX via Voicepulse or others I know that I can simply have one registration statement along with an inbound context, then in extension.conf map the outbound context. from iax.conf: register

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Hello, I have asterisk setup with Broadvoice. It works great as PBX and Outgoing calling server for all local clients withing 192.168.1.0 network. My asterisk is running over NAT. I use linksys router. Now, I am trying to connect from outside to my asterisk server. I use Diax as iax client. For some reason I cannot connect to my server from outside. On my router I forward those ports to my

We currently have an Asterisk server set-up, serving a handful of sip-phones and sipuras, and connecting to the outside world via one FXO and various SIP and IAX providers. In order to conserve bandwidth and have a fall-back in case our internet connection becomes unavailable, we're looking at putting * on a hosted server and funnel the calls to our office, such that: DID DID DID

I have been picking at Asterisk for about a week, and I think I'm close. I was hoping for a little guidance to bring this on home. I want to be able to make outgoing calls from my SJPhone clients using my VoicePulse Connect account. I have the two requisite items from Voice Pulse, but I've had no luck successfully integrating the VoicePulse settings into iax.conf. My current config:

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

Hi. I'm new here so I hope this is a sensible question/sensible place for it. I have a PSTN to IAX phone number with voipuser.org that I'm using to test an IVR service. The only trouble is that after approximately 40 seconds of silence (e.g. after background playback of a menu prompt) the call gets cut off. Is this a common problem? I've already set the ResponseTimeout to a big

I have an IAX hardphone connected to an Asterisk appliance sending and receiving calls via IAX to three different providers. The appliance is currently connected to a NAT router. The appliance is purposely being set up via the GUI, not in messing with any config directly. One of the "service providers" is my other asterisk box on a different Internet connection. In all cases, outgoing

I'm very, very confused. Dialing out, through VoicePulse, with both gsm and ulaw CODECs, most of my calls are great. However, calling my (non-Asterisk) voicemail at my job, and calling my cell phone both produce horrendous (~ 1/3-second delay) echo. I've tried with different phones (Polycom and Grandstream), different IAX CODECs (as described, above), different network