similar to: Newbie Issues => SIP won't stay connected, and IAX Unable to Create Channel

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

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

Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and

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

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

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

Does anybody know of any commercial providers of IAX termination with DIDs in the Seattle, WA area? I believe the area codes are: 425, 206, 253 Failing any commercial providers, is there anybody in the seattle area running Asterisk with a PRI coming in who might be willing to sell me an IAX trunk with a DID in Seattle? -- ____________________________________________________________ Muiz

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

Dear All: I have several boxes that up and running just great, then we changed internet equipment due to a lightning strike, now all my inbound IAX connections (iax2 show peers) have unknown status. If I log into the remote boxes, it says "Request sent." The authentications haven't changed at all, and all the iax.conf settings are correct. It looks like a firewall issue, but

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

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

Just a quick couple of questions for ya'll. 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones? I have had a terrible time getting a hold of anyone over there, and I need this functionality before I can migrate to * completely. 2) Are there currently any problems with inbound DID's? Everything is setup properly in *, but I am not able to receive inbound calls,

I have working SIP numbers with broadvoice, and just added a DID from http://connect.voicepulse.com/ . The calls answer, but DTMF is not recognized. With "iax2 debug" active pressing DTMF does nothing. Zilch. Zero. A friend tried a different IAX2 connection, and got the same results. I see the following in the archives: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: > Hey

I have tried for over a month off and on to get iaxtel for inbound to work... and tonight after alot of troubleshooting we noticed this: iaxtel inbound will use the last entry in your iax.conf to auth against. So if [iaxtel] is at the top and say [voicepulse] at the bottom. An inbound call will try to auth against that [voicepulse] entry even with the [iaxtel] entry at the top of the file. Has

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

I've been beating my head for 5 hours to figure out why my asterisk server or sipura isn't passing my voice over to the caller. It seems i can hear the caller but they can't hear me it seems either the asterisk or the sipura isn't passing this information. Here's my setup specs asterisk server 0.7.1 - X100P Card - Sipura 2000 - Nufone Service - Voicepulse

I am having probelms connecting to voicepulse this morning. Is anybody else having issues.. burak

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,

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)