similar to: Odd problem with sip.conf register command:

Okay, my last post for the night. So, after tweaking my iptables rules I placed a test call from my * -> Vonage softphone account, to my Vonage main account (which uses the Motorola device). Once connected I tried punching a few DTMF's to check the quality and found I had been transferred to a fax machine. ???? Did I find a secret entrance to some fax machine at Vonage, or

I've got a Linksys PAP2 on my Vonage account with unlimited usage, but my softphone-addon account only has 500 minutes. Anyone ever try to setup their * to mimick the Linksys PAP2 ? Any comments or suggestions on what problems I might run into if I try? -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've yet to successfully register and receive calls from my Vonage softphone. I've tried what few examples are given in former post this list and some other forums, but nobody seems to be stepping forward saying it works recently. Either they broke something to where you simply cannot use *, or the config examples need updated. If anyone can show a still working config,

Does anyone have experience setting up * to accept anonymous sip UAs and the dumping the call into IVR? I'm thinking this would be a good way to have customers call us without creating an extension. So for my tests have been focused on providing internal functionality. Thanks, Chad -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm trying to get Asterisk to connect to Vonage (softphone acct) to allow me to place and receive calls. I have successfully configured Asterisk to route inbound calls and send them to the correct extension, but I can't get outbound calls to work. I have Asterisk successfully registering with Vonage, but when an INVITE is sent out, I get a "404 Not Found" back from Vonage

Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. ? Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA

Hi All, I am trying to setup another sip trunk in addition to what I am already using. The sip provider we are using right now gives you your username as your email address. So IE. If my email is james@james.com.... that is my username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate

Hi All I am trying to dial out using SIP and Vonage using the instructions : <a href="http&#58;&#47;&#47;www.voip-info.org&#47;wiki&#47;view&#47;Asterisk&#43;and&#43;Vonage" target="_blank"

i have tried to connect my asterisk server to vonage like this: Sip.conf: register => 1<yournumber>:<secret>@atlas-east.vonage.net:5060 [vonage] type=friend username=1<yournumber> secret=<secret> host=atlas-east.vonage.net port=5060 allow=all maxexpirey=15 dtmfmode=inband fromuser=1<yournumber> fromdomain=atlas-east.vonage.net canreinvite=no nat=yes

I easily get confused when try to undertstand FXO & FXS ports. Is it possible to use an ATA to connect to a TDM400 card. If so, would I use FXO modules or FXS modules ? My goal is to connect my asterisk server to Vonage (via the ATA they send me) so I can use thier standard plan and do with out the Softphone account feature that only allows a few hundred minutes talk time. Thank you, Steve

I setup a vonage account last year, and cancelled it last night when I put my asterisk box together and signed up for a Broadvoice account to use with it. Now I would like to use my Linksys router as an MTA, but realize it is still programmed with all of vonage's proprietary information and I do not know how to clear it. I understand that just pushing the reset button will not do it.

On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota <murthy64 at hotmail.com> wrote: > Tested with X-Lite and it worked fiine. Is there some way to replace > "Anonymous" with a config parameter? > > Thanks for your kind help > > ---------------------------------------- > > From: murthy64 at hotmail.com > > To: asterisk-users at lists.digium.com >

I just got setup with vonage.ca with the motorola ata unit.. I fired up ethreal and checked out what's flying over the network... The sniff below would lead me to believe that it might be possible to have asterisk spoof the User-Agent field and register itself? Any thoughts/feedback? Thanks. > > No. Time Source Destination Protocol Info >

I just installed Asterisk on my server and I have Vonage softphone. I need my Asterisk server to receive calls through the Vonage Softphone DID and make outgoing calls through the Vonage ATA using an X100p to connect to it. Can someone help me out on configuring this? I really need this for my business and would greatly appreciate the help.

So far in my playing with Asterisk I've messed with soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters (Grandstream 286, Digium IAXy). I've also got a Vonage line, using a Linksys ATA. None of the devices I've connected to my Asterisk server have been able to maintain the same consistent sound quality over a long distance as the Vonage line. Don't

Will Vonage unlock your ATA for you for the $15? Or someone else? I have an ATA from them I would like to use with asterisk as well. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Thursday, April 29, 2004 8:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Vonage and * (and what

After over a month (well, ok, no more than an hour a day :) of planning, getting hardware, tinkering and testing, I'm about to my Ultimate Home Phone System (tm) online. Connectivity to the outside world is provided by: A. 1 POTS phone line connected through an X100P ($11/month, needed to carry DSL) B. 1 Vonage ATA186 connected through an X100P (needed for the rate center :( ) C. 3 Broadvoice

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

There was a flurry of "Vonage is going to unlock SIP" activity last year; did anything productive ever come of it? Are *you* using your Vonage lines directly into Asterisk? In lieu of that, for a 4 line small business that doesn't need to pay Vonage $150 a month, who? Broadvoice? Someone else? I'm a touch unimpressed with the fact that BV's website *won't quote you