similar to: read .what else to do ?

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

I have d/l the iso (*@home 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to

Guys. Im writing this because Ive checked the wiki, Xten website and read a lot of docs and still cant figure out a way around the NAT issues. Maybe somebody else can give me some ideas from a fresh perpective. My test setup is this: Asterisk -> 2wire homeportal Firewall -> internet Computer with Xten eyebeam The asterisk box and the computer with xten beam are behind the same

Hi, I'm experimenting with the dev kit lite and now past the USB unpleasantness it's working great with standard phones and lines. The priority right now is getting soft phones (under Windows XP) working well. So far, I've only been able to get the XTEN Lite phone working and I really don't understand how I set it up. I used "xten" for every option everywhere (display

Hi Everyone, I'm new to * and I'm trying to setup a small configuration of SIP clients. Eventually when I get this working I plan on expanding with a Digium developers kit to add analog phones and PSTN access. My two end points are an Xten softphone and a Mitel 5055 SIP phone. Both peers seem to register with * but I cannot call to one another. When I dial the associated extension, the

My question: How do I configure AAH via AMP to make a connection through our legacy PBX to the PSTN? Details: We're trying out Asterisk through Asterisk @ Home. Our legacy PBX has a modem type dial tone port that we hooked a Digium FXO to. Now I can dial from the XTEN client on my computer to any legacy PBX extension. If I connect a regular phone to the modem dial tone port, I can dial

Hi folks. Thanks for having patience with a newbie... I am going through the setup exercise at http://chayden.net/Asterisk/SeUpAsteriskAtHome.htm I am trying to do it all on BroadVoice since this is just experimental, and the 100 minutes of outgoing from them should be enough for now. So I set up BroadVoice for both inbound and outbound, using a combination of the instructions referenced

Hi all, Seriously, I've tried to read everything I could find (& search for) on voip-info.org and other sites about this problem, but have been unsuccesful. Equipment: xten lite X100P Whitebox linux running Asterisk / AMP D-Link DI-804HV (VPN router) I have installed another DI-804HV at a second location and created a tunnel. For the computers behind that unit, everything works fine

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

Hi: We have got SIP clients connecting to our Asterisk fine with a DSL connection behind router (NAT), but when we bring the Sipura 2000 ATA to a Rogers Cable connection behind a Netgear router (NAT), the SIP clients aren't able to reach the Asterisk at all. We enabled the SIP debug in Asterisk, and it doesn't see any request coming from these SIP clients, and we also tried the to use a

Hello Folks- Pretty new to the list here, got a lot of reading to do.. Does anyone know where I can find a decent HOWTO or set of instructions for running Asterisk and SIP clients thru firewall/NAT systems? I have a Asterisk box sitting behind a linux firewall at a remote location and have the 5060 and etc ports open as well at 16381-16391 UDP open and routed to the Asterisk box as well. I have

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco

Hi all, This is the first time i am using stun with asterisk for nat problems. I have read the rfc which describes how stun works. i didnt have any problems understanding it. I have also intalled the stun server called stund which i downloaded from sourceforge. I have seen on the list that most people use stund here. I have started the stun server and its running silently. Now i dont know what to

Am I the only person in the * world who can't get a sip connection through an iptables firewall? I've got everything else working fine. Xten <-> PSTN, Xten <-> Analog, IAX <-> IAX, but exten => 3733,1,Dial(SIP/fred@somewhere.com) ; evades me, ngrep @ port 5060 says the INVITES go out but how do I get something back? -- Dave Cotton <dcotton@linuxautrement.com>

Running * with default config files except for sip.conf. Any call made is dropped 5 seconds after connection, with the following messages: Mar 17 16:37:41 WARNING[1009461760]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call 6C94C1B1-77C4-11D8-91FB- 000A95DA04DA@192.168.1.152 for seqno 48221 (Response) == Spawn extension (default, s, 5) exited non-zero on 'SIP/2000-6bd7' Mar

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

So it was all working well and then suddenly I'm unable to get incoming calls from BV. Outgoing is fine. I'm using AAH. I have the following settings; register=9738281625@sip.broadvoice.com:PASSWORD-GOES-HERE:9738281625@sip.broadvoice.com/2208 [broadvoice] username=9738281625 user=phone type=peer secret=PASSWORD-GOES-HERE qualify=1000 port=5060 nat=yes insecure=very

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

Hi all, I have a quick question I hoping someone can help me with. I have Asterisk@Home running and working just fine. I've integrated it with BroadVoice and so far I'm blown away by everything I can do. I don't particularly like sitting my entire machine in the DMZ on my network sitting open, but when I do I can run XTen on my PC at work and make/take calls with no problems. So I