similar to: VoicePulse Connect 4 Channel Limit?

I've been using my voicepulse connect number for over a month now, but today it simply won't connect. My partner and I each have a number, both are mapped in my iax.conf and extensions.conf files. This has been working fine. Today, either number gives this message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request

I am trying to set up IAX with Voicepulse. When I turn on debugging I get the following message when I call my PSTN number: NOTICE[1142106560]: File chan_iax2.c, Line 4321 (socket_read): Rejected connect attempt from 66.234.228.132, request '3017275115@VPWS' does not exist Any help would be GREATLY appreciated. Thanks, Isaac isaacmcdonald@attbi.com

G'Day List, Can someone help me out a bit please. I just installed Fedora Core 3 on a Dell Power Edge 400 SC and am trying to install * I am following http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation After running: cd /usr/src/asterisk make clean make make install make samples The instructions says: Configuring Asterisk - Login to your server as user

Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK

Hi, First i would like to thank the kind people of the list who have answered my previuos mail, but i am still stuck as asterisk still crashes upon startup, i have read the install article at http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation and i have search the asterisk archives, but i still cant get asterisk to work, i have tried reinstalling asterisk but it still complains and

I'm trying to use Jeremy's suggestion of dialling using just the peer name instead of user:pass@peer but I'm running into some really funky issues. It does the same thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows

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 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

[Apr 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -

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 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

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 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.

Hey I just setup service with IPComms and they are telling me to setup such as this: iax.conf: [IPCommsNet] type=user host=69.15.xxx.xx context=voicepulse-in ;(changed by me) nat=yes dtmfmode=rfc2833 disallow=all allow=ulaw allow=alaw allow=gsm When I'm calling once of my numbers it's giving me this though: Oct 5 12:11:06 NOTICE[49584]: chan_iax2.c:5476 socket_read: Rejected connect

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

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"

I'm having trouble configuring these services the way I want. Basically I prefer using iLBC before GSM, however Iaxtel only want to talk GSM. It _seems_ that Voicepulse prefers GSM also, because even if I put ILBC before GSM in the "allow=" part of iax.conf, if GSM is mentioned, Voicepulse will use it. If I don't allow GSM Voicepulse will use ILBC. Does anyone know how to

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl