similar to: Registering multiple FWD accounts

I run an Apple OS X workstation and I've got a server on the same LAN that's both a webserver and an Asterisk PBX. I wanted to be able to originate calls in the OS X Address Book application, and have Asterisk dial them and connect them to the phone on my desk. I've assembled a system that uses AppleScript to connect, via XML-RPC, to a web application that, in turn, connects to

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

Hey all, I'm trying to get my Asterisk server up and running on fwd.pulver.com just to get the hang of it until I get my FXO card in a couple of days. It seems to connect but that's about it. If I try to dial into it from another fwd # it says user is not online. In sip.conf I have the following added: register => xxxxxx:xxxxxx@fwd.pulver.com/489125 [fwd.pulver.com] type=friend

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hi All, Is there any way to connect (register, initiate and receive calls) with Asterisk to FWD through NAT? Since I own my router port forwarding is not a problem. I tried with Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwd.pulver.com but since Asterisk still use internal IP in some SIP fields I got "479 We don't accept private IP contacts. Please set your external

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

Sirs, Thank you for your quick response. But when i try to make a call to FWD the following error appears: For example, when i call to 612 (a service number of FWD) -- Executing Dial("SIP/Phone4-e85b", "SIP/612@fwd.pulver.com|90|Ttr") in new stack -- Called 612@fwd.pulver.com -- Got SIP response 500 "I'm terribly sorry, server error occured (1/SL)"

I bought three plus two Grandstream BudgeTone 101 phones. The shipping cost more than the phone itself from Pulver store. The first shipping had one phone defect. Nothing on the display. (Can happen!) The second shipment had one phone with a defect display, but it still worked. The second phone's handset was defect too (microphone did not work). Changing the handset from this one to the

I've been having issues getting FWD to work. I posted this same Q to the FWD forum (no responses yet), but I was hoping someone here had some insight: http://yabb.pulver.com/cgi-bin/yabb/YaBB.cgi?board=news;action=display;num=1072263468;start=0#0 I just signed up for an FWD account (I know I had one before, but I lost it.. :) I've got it running through Asterisk - all working fine

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Hi, I have some doubts on sip.conf. 1) Can I have two or more SIP phones acting as extensions in one Asterisk box, and at the same time, registered to a SIP proxy, say Free World Dialup? If yes, how? 2) Why we need a section in the sip.conf for the proxy, say, Free World Dialup's fwd.pulver.com? In the case of 1), how to assign the value to section [fwd.pulver.com], since there are more

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if

I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of

I have successfully configured asterisk to make outgoing calls over FWD, but cannot receive incoming calls. The console shows no messages, even though an XTEN client on the same network has no problems receiving incoming calls. This is the relevant part of sip.conf [general] ..... register => 688426:xxxxxxxx@fwd.pulver.com/6000 [fwd.pulver.com] type=friend username=688426 fromuser=688426

Thanks for taking a look greg and hank. This seems to be getting bettre everyday..help please My sjphone is running on the same box as asterisk...i believe then the red hat firewall should not be a problem. Whenever i dial from CLI i get ######### Executing Goto("OSS/dsp", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("OSS/dsp",

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>