similar to: asterisk and openvpn and sip

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi, I am running OpenVPN where i have one central hub VPN server, and multiple spoke VPN clients. I can ping from each client to the server and each client to computers on the subnet which the server resides (192.168.2.0/24) so it works ok there. I cannot however, ping from one client to another client. I guess the packet path would go: clienta -> vpn -> shorewall/router -> vpn ->

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

I'm using sipsak to send messages to Snoms in my subnet. At work, works fine: sipsak -M -O desktop -B "foo" -s sip:1001@192.168.1.220 -H 192.168.1.46 displays "foo" on the Snom display On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no VLAN, no routing) the same command (modified for my LAN) always yields: (type: 3, code: 3): from 192.168.171.8 at the console

Hi, I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P (12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I connected an analog phone in parallel to make a test: __________Asterisk fxo ---- line -----| -----------------Analog phone The analog phone rings immediately when calling, while asterisk shows the message

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP

hey friends, I have installed OpenVPN 2.0.7 (i386-redhat-linux-gnu [SSL] [LZO] [EPOLL] built on Apr 29 2006) on Centos4.0 through rpm (diag repository). The network scenario of my office is below Remote Client ----> Internet <-------> Cisco Pix Firewall (Gateway) <----> VPN Server & LAN Clients

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

Hi all, I have upgraded my Asterisk box from 1.2.x to 1.4.x version: it seems that sometimes some phones become paused and cannot receive calls anymore. I tried to set autopause = no in every section of my queues.conf but nothing changes.... Anybody knows why a phone becomes paused? Is it an Asterisk 1.4 bug or there is a particular reason for this behaviour? Thank you. Giorgio.

Hello, I want to create a VPN between two Asterisk servers using OpenVPN. How to configure Asterisk and OpenVPN to do that. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070809/ddaed76b/attachment.htm

Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

You definetly need a TAP connection to make samba work over VPN. We use OpenVPN host2net-accounts created with IPCop here since quite a while and it works with samba without problems. However, the speed is of course not as fast as in local net, but this is rather related to the way the SMB-protocoll works. ;-) ________________________________________ Von: samba [samba-bounces at

Hi all, I''m trying to use OpenVPN as a VPN solution on a firewall running Shorewall. The IPSEC VPN I tried first has shown a little bit unstable under several conditions, especially with Windows clients. As OpenVPN is best run in ''bridged'' mode (see http://fedoranews.org/contributors/florin_andrei/openvpn/), I became interested in the bridge capabilities of

Hello, I have the need to connect 2 remote site with vpn, the windows pc of the 2 site it can share the HD and printer. This is my configuration : LOCAL NETWORK A : ip from 192.168.10.2 to 192.168.10.99 | | | | eth0: 192.168.10.1 FIREWALL A : ( with debian ; openvpn ver. 2.0.beta15 ; shorewall ver 2.0.11 ) eth1 : xxx.xxx.xxx.xxx ( pubblic ip address ) | | | | INTERNET | | | eth1 :

Hello all, Let's say I have an OpenVPN (v2) server sitting on a Linux machine with the IP address of, say, 192.168.10.1o. We are talking real address, assigned to a NIC on the machine. Now let us say the OpenVPN server hands out IP's in the 192.168.20.0/24range. And let us say that I want the machines able to reach the VPN server to be able to route to the machines available via the VPN.