similar to: Unlocking Vonage ATA 186

Don't know if this has been addressed, but why isn't there a file_include style directive for extensions.conf? I find that my extensions.conf grows a lot, and it would be a lot nicer to have a tree of files rather than one big file to try and navigate. Also, I've got a couple different 'systems' running concurrently on one asterisk box (ie, completely different groups of

I've been using * CVS code from May of this year and was able to connect to iConnectHere and receive calls with * being behind NAT. Now that I've upgraded to 1.0 RC1, this no longer works. I've tried setting nat=yes in places, externip, et al with no success .. even though the code I was running from back then worked without that. Any suggestions? BTW, I've gotten DTMF from

Hello, today I found this security report regarding Asterisk SIP Security. http://www.securiteam.com/securitynews/5LP0720B5G.html Maybe It could help somebody who isn't using a newer than 15th of August cvs version. Best regards Lubo

Hello, There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html This is the second security report regarding asterisk for 8 days (http://www.securiteam.com/securitynews/5LP0720B5G.html) Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list

I just signed up for Voicepulse with a DID. I can register with Voicepulse and dialout just fine. Only problem is that when I dial my DID from my POTS line I just get a fast busy and nothing in the console. Any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040104/d5bd5bd3/attachment.htm

If anyone out there has an ATA-186 that they purchased but cannot use with Asterisk due to it's being locked by Vonage, please contact me off-list. JT

I am a game developer, and we used RC3 in our last game. Worked great. I'm trying to update to 1.0 now, and running into some problems. Specifically, I think a change to _get_prev_page() in vorbisfile.c broke the decoder for very small files. I have a file called "silence" that's just a silent wav file. It compresses down to barely more than the header, so it's _really_

R-geographers... I'm trying to solve a problem to implement a line-by-line tiled processing using RGDAL (read 1 line of an image, process the one line, write one line of the image to a binary file). Everything except for the final step I'm able to do using a combination of RGDAL and r-base commands. Below is the basic structure, the input file "elev" can be any image file

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.

In http://www.voip-info.org/wiki-Asterisk+agents as followings, what type of channel of 28 and 29 is? agents.conf [agents] agent => 1001,4321,Wayne Kerr queues.conf [queue1] member => Agent/1001 extensions.conf exten => 28,1,AgentLogin(1001) exten => 29,1,Queue(queue1) I use the following in extension.conf with Vonage softphone account, it works well to call

I've been reading all I can in order to try to implement an asterisk setup. In speaking with someone the other day, they advised they thought there was a way to make a SIP or softphone (gnophone) go straight out to the vonage network through asterisk. I already have vonage set up with an ATA 186. What I'm wondering is is there some way that I could direct dial from a softphone

The Vonage service is offered with a SIP Cisco ATA device for connection to an analog phone. Is it possible to connect the Vonage service directly to the Asterisk PBX bypassing the ATA and FXO card? Are there other services that offer this capability or something similar to IP dialtone? Thanks Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL:

> -----Original Message----- > From: Asterisk@gtcus.com [mailto:Asterisk@gtcus.com] > Sent: Saturday, September 06, 2003 8:39 AM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] VONAGE or IP Dialtone > > > Thanks for the great feedback on these options. I am fairly > new at this and not familiar with the IAX/IAX2 capabilities > offered by

In taking a cursory browse at Vonage's site today, I noticed they are now offering a soft phone. Has anyone had any experience using this? And does this possibly open new opportunities for using Vonage with Asterisk? Just thinking outloud on the list, soliciting thoughts and experiences from others. AJ

I got a call from an ISP friend tonight who said he is getting calls from people who are getting signed up with Vonage. Instead of sending them ATA186s, apparently they're receiving something made by Motorola. They apparently work significantly differently than the Cisco units, and there have been some problems. Anybody know anything further? Thx. B.

FYI http://www.itbusiness.ca/index.asp?theaction=61&sid=55298 did not like this by line in the story t"he CRTC has said it will likely regulate voice over IP the same as other phone services."

Maybe somebody can help me. I'm testing my brand new [*], interconnecting it with several networks, but only via softphones up to now. I do have a VT1000 CPE from vonage, with a vonage line on the first FXS port, and would like to use the second FXS port to connect it to my [*] This is my question: My vonage VT1000 is connected between the cablemodem and a home gateway. Therefore, I

My previous post hasn't even made it to the list yet (am I being moderated?), but I got Vonage's Softphone service working for inbound and outbound calls. Keep in mind that there's currently no perceived limit on simultaneous inbound calls, which makes this a wonderful solution for Asterisk (at least for my use). Below is a sanitized snippet from my working sip.conf; your mileage may

atlast after working of 7 hours i got voange soft account working on asterisk.

...now, we don't wish them ill, but Vonage seems to have been out of commission for quite a little while. Website is excruciatingly slow, log-in fails, hard-line and soft-line are out (alert-tone or "subscriber not in service") -- even "Network failover" is failing. Outgoing calls return fast-busy. 866-support number is busy. Let's see how they deal with that in