similar to: dialling peer problems

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

hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am

In response to the postings by Andrew Kohlsmith and Ernest W. Lessenger: Andrew, I modified the exten line in extensions.conf as you suggested. Unfortunately, It still does not work... Ernest, I spent approx. 4 hours reading list archives (and anything else Google served up) on how to configure iax.conf and extensions.conf to work with Voicepulse. Then, I sent an email to voicepulse

Greetings, I have been experimenting with Asterisk for a few weeks and finally decided to take the plunge and purchase a few DIDs for inbound calling. Our attempts at IAX/IAX2 connectivity with VoicePulse have been less than successful. We get "Registration Refused" errors from Asterisk whenever we launch the server. The front-line support folks at VoicePulse suggested that we are

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

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'm trying to set up asterisk so any phone connected to channel 1-16 of my Adit600 channel bank can pick up a call coming in on channel 24. I do not wish to ring any of the 16 channels on an incoming call -- this is strictly so I can pick up the line if I see it ringing and wish to answer at work. I have channel 24 in call group 3, and channels 1-16 in pickup groups 1 and 3. However

The problem: T100P connected to an Adit600. Channel 1-16 are FXS, 17-24 FXO. I have Zap/24 in callgroup 3 and Zap/1-16 in pickupgroup 3. When a call comes in on Zap/24 I cannot pick it up with *8 from Zap/1-16. *CLI> show version Asterisk CVS-04/27/04-23:48:08 built by root@tuck on a i686 running Linux The zapata.conf and extensions.conf are located here:

Hello. The procedure so that it works you can find in: http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris k a the files .wav chmod 755 file.wav sox file.wav -r 8000 file.gsm resample -ql chmod 755 file.gsm in extensions.conf xxxx=> xxx,x,playback(file) Ing Javier Rios Ing de Proyectos 04167285748 212 2637246 /2637187 -----Original Message----- From:

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Is the channel physically being hung up before the * tone is heard? Good question. If it is, Asterisk doesn't detect it -- the PBX doesn't support Kewlstart-style disconnect notification. The sequence I hear on the extension, when I plug in an analog phone, is the click of the

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've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

> -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Tuesday, 25 November, 2003 08:56 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Picking a channel (FXO port or SIP) for outbound calls > > > > Yep, we use it for international calling. Works great: > > exten =>

At 1/16/04 7:25 AM, Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: >That's pure bullshit -- I use software RAID *specifically* because I value >my data. I don't want to buy two hardaware RAID controllers to have one >sit on the shelf just in case the first dies... and if the second dies >you're SOL because they've lasted long enough that

On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:akohlsmith-asterisk@benshaw.com] wrote: > On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: > > Received: from source ([81.56.129.44]) by exprod5mx8.postini.com > > ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT > > > > Your MTA claimed it was called "SOURCE" but rDNS tells the recipient

On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote: > Do you find it difficult to manage four LD providers? > Can you show me part of your LD Macro and how it's used? > > I'm toying with two LD providers now, but don't have failover setup. >Just using each one for what they are best at (least cost). > >Thanks, >--Ben Kirkpatrick > > Not

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

RE Message: 5 Date: Sat, 7 May 2005 23:18:46 -0400 From: Andrew Kohlsmith akohlsmith-asterisk@benshaw.com Subject: Re: [Asterisk-Users] At home Asterisk via Broadvoice? To: asterisk-users@lists.digium.com On May 7, 2005 11:04 pm, John Stegenga wrote: > Broadvoice will give me 2 lines, with 2 phone numbers each - distinctive > ring - for a reasonable fee... Please do a google search for

All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk. On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a

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"