similar to: Configuring server to call SIP numbers on the Net?

Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as "sip:user at domain.com", such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which

Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.

Hello I don't have a landine and use a VOSP to provide access to the telephone network. In case a call comes in and I'm not home, I'd like Asterisk to log the call, and then send an SIP message to my VOSP so the call is forwarded to my cellphone and is thus charged to the caller, without Asterisk having to dial out to my cellphone through my VOSP at my expense and bridge the two

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite calls out the remote XLite, everything works fine;

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hello I read a whole book on OpenVPN, but still can't figure how to configure the server + client so that the the client connects and sends SIP/RTP data through the tunnel. To get started, I'd rather use a shared key instead of X509 (certificates + keys). The server is running on a uClinux appliance, with /dev/net/tun, and OpenVPN is 2.0.9. The clients will be Windows hosts connecting

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with Asterisk. How can I tell

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Hello I just set up Asterisk 1.6.2.9 through packages on a test host running Ubuntu 11.04, configured sip.conf/extensions.conf, and launched EyeBeam 1.5.20 to run the echo test. For some reason, even through I'm using a headset, there's a lot of echo and after a few seconds, it sounds like it enters a very fast loop before the echo stops somewhat. IOW, unusable sound. Here's a

Hello For users who 1) don't have a QoS-capable ADSL router and 2) would like to run Asterisk with a couple of SIP trunks, I was wondering what hardware is recommend to run any of the main open-source *WRT projects to which Asterisk has been ported: (http://en.wikipedia.org/wiki/List_of_wireless_router_firmware_projects Thank you.

Hello I thought I had things set OK to have Asterisk play FR files for prompts and MOH, but for some reason, it still can't find them: ============ ll /var/lib/asterisk/sounds/ drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/ drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/ Note: fr/ contains core + extra + moh as downloaded from here:

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello I was wondering if someone knew of an application that could check that the user has a firewall and a broadband connection that will work OK with Asterisk and VoIP. The app would first perform some bandwith + jitter tests, and will then call a STUN server to check that the firewall isn't symetric. BTW, is Asterisk now STUN-capable, or is it still to map ports manually on the firewall

Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the

I have asterisk up and running on a fedora system but having trouble accessing system via softphone (ekiga and xlite). Im a newbie to asterisk and was looking for some help walking through this. I imagine 10 - 15 mins would be all needed to make proper config changes needed. Once I get this setup I'd be interested in discussing customizations and scripts so any freelancers or companies welcome

Hello I've done some googling, but still puzzled at my working configuration. Apparently, a "user" can only receive calls through Asterisk, a "peer" can only make calls, and a "friend" can do both. If that's correct, I don't understand why my VOSP requires the following settings in sip.conf to let my Asterisk server make/receive calls to/from the PSTN:

Hello, I need to write a script which prompts the callee to type a number, and then read it back to them as confirmation: ======= extensions.conf [robocall] ;Expect 10-digit number excluding final #, 2 tries, 20s time-out exten => s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20) exten => s,n,GotoIf($[${LEN(${NBR2CALL})} != 10]?end) ;exten => s,n,SayDigits(${NBR2CALL}) exten

Hello I'd like to configure Asterisk so that... 1. I ring it from my cellphone with CID number displayed, just to notify Asterisk that I wish to make a call 2. Asterisk waits until I hang up, calls me back, and prompts me for the number I wish to call 3. Asterisk puts me on hold through Flash(), which is apparently the equivalent of hitting the R key on European handsets 4. Asterisk calls the

Hello I'm sure someone has already tried this: I use a couple of electric heaters to heat my office. I'd like to somehow connect them to Asterisk so that I could switch them on remotely by either calling the IVR or sending an e-mail to the Asterisk host, so that the room is warm when I get to the office :-) Any information appreciated. Thank you.