similar to: Getting VoicePulse # listed in 411

Hi all, I'm working on a setup for a small office. I'd like to use SIP/iconnecthere most of the time, because they're cheap. But they only allow a single call. When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse instead: exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN} Well,

Are there other providers similiar to VoicePulse-Connect? I signed up and added a $30 credit with them, only to later find out they don't have coverage in the 757 NPA. So hopefully they will refund the $$ since otherwise I won't ever use them since my goal is incoming only, and #2 I'm still without a 757 dialup. I would actually like to find a service that will let me really have the

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Hello, I'm planning to convert my phone system to Asterisk, as I've outgrown my TalkSwitch system. I have a few questions for experienced * users, most of which can be answered yes/no. Current Setup: - Talkswitch 48NLS (4CO/8Ext) phone system. - One CO line, two Vonage lines, one Voicepulse line connected to phone system - A third Vonage line directly connected to a fax machine - A

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 didn't know if it would work or not, but I figured I'd try slow-speed half-duplex TDD over GSM & Vonage. I called a AGI script I have that speaks to TTYs, by calling from Vonage to one of my Voicepulse lines. I don't control the Vonage codec, so I have no idea what it uses, but I am using GSM for the Voicepulse line. Everything worked fine - echo canceling didn't cause any

I want to collocate an * box somewhere, where better than where voicepulse chose to put their servers? They probably did their homework and selected someplace where good handoff to the pstn can be found, right/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050520/9f5975b8/attachment.htm

I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com.

Until recently, I was using Broadvoice for my in/out calling thru Asterisk. I was extremely pleased to see that Broadvoice was actually passing the callerid info (number and text) that I had set up on each device in my SIP.CONF file. I had PSTN users tell me that they were actually seeing name and extension info when I called them from the Asterisk box. Last week, due to numerous user quality

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

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

For this week's conference, the two founders of Voicepulse, Ravi Sakaria and Ketan Patel, will be joining us. For those of you who are not aware, Voicepulse is an asterisk friendly VOIP provider that has won awards for service and innovation. We will also have Trixbox news, updates, as well as discount codes. Lastly, we are working feverishly to bring you more information regarding legal

Voicepulse has ignored four emails over the course of two weeks. Anyone have any ideas of whats wrong? - Executing Dial("IAX2/voicepulse-in-01@66.234.228.170:4569/7", "IAX2/acctname:acctpass@gwiaxt01.voicepulse.com/14109649073") in new stack -- Called acctname:acctpass@gwiaxt01.voicepulse.com/14109649073 Sep 13 22:48:25 WARNING[131080]: chan_iax2.c:5375

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

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,

Okay, I can receive calls through Voicepulse fine. All the various attempts (too many to list) to create a workable configuration to Dial to Voicepulse has failed, from 403s to "No authority found" to nothing. The Voicepulse folks told me that the open access was SIP and I shouldn't have a reference in the iax.conf file, but then said that they were refering my question to the

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

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

Greetings, I am new to all this VoIP stuff and have been having a bit of a hard time getting my soft phone working as a SIP client thru Asterisk. I apologize to start off with such a simple question and hope it's ok to post this and see what others have done. THE GOOD NEWS: I have successfully setup Asterisk 1.07 on an OSX machine. The build is running and working successfully. I am able

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