Re: [asterisk-users] odd issue with the with SIP over VPN
I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client -- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? You may have run into a problem I've encountered with the SIP client in the N810, or something related to it. One of the complexities/weaknesses of the SIP protocol, is that each SIP node puts its own IP address into the protocol packets it sends to its peer. The peer uses this IP address (embedded in the SIP headers) rather than the IP address in the actual IP headers, to manage the conversation. This means that each SIP peer needs to know what IP address to announce and it has to be an IP address which is usable to the peer, or the protocol won't work. The SIP client on the N810 (and the N900 I imagine) will, under normal circumstances, *always* specify the IP address of the main IP interface (wireless). This happens even if the call is being routed through a VPN. Or, if you have STUN support turned on, it may specify whatever IP address the system deduces is the visible public IP address of whatever NAT it's living behind. Neither of these IP addresses is likely to be usable, if the conversation is taking place through a VPN. What you probably want, in this case, is for the SIP packets to contain the N900's VPN endpoint address. The Maemo SIP client doesn't know how to do this, at least not without assistance. Fortunately, it's possible to assist the client, via some scripts or push-commands in your OpenVPN configuration. The methods differ a bit depending on whether one is running OS 2008 (Diablo) on the N810, or Fremantle on the N900, but the principle is the same. See https://bugs.maemo.org/show_bug.cgi?id=1860 for a discussion of the problem, and for some sample scripts. I've been using this approach with my N180, and (via manual configuration) with a Linux laptop with OpenVPN and Twinkle. It works fine, and seems more reliable than trying to use STUN. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd issue with the with SIP over VPN
On Sun, 24 Jan 2010 10:30:13 -0800 Dave Platt dpl...@radagast.org wrote: snip Awesome. This client bug was the cause of this. Much thanks. signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] odd issue with the with SIP over VPN
I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client -- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? signature.asc Description: PGP signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd issue with the with SIP over VPN
You're going to be a lot more specific about the precise - if symptomatic - meaning of do not. On 01/23/2010 09:08 PM, Zane C.B. wrote: I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client-- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -- Alex Balashov - Principal Evariste Systems LLC Tel: +1 678-954-0670 Direct : +1 678-954-0671 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd issue with the with SIP over VPN
You probably are not advertising the routes across the vpn properly. Does your setup look like this asterisk[network a]openVPN server[network b - vpn]-openVPN client[network c]-sip client where network a, b, and c are all separate subnets? Is your vpn setup for routing or bridging? you need to make sure that the vpn server allows network a to to talk to network c, and that network c can talk to network a. By default, only the client can talk to the server. Attached subnets will not be routed automatically. See this section of the openVPN howto: http://openvpn.net/index.php/open-source/documentation/howto.html#scope I have 3 asterisk servers with sip trunking between them all running over openVPN links, and everything works fine when you make sure you setup the routing right in the vpn. I also have 2 phones that connect to one of the servers over a openVPN link as well - they're not sip (Nortel unistim) but it also works just fine. Andrew On Sat, Jan 23, 2010 at 7:17 PM, Alex Balashov abalas...@evaristesys.com wrote: You're going to be a lot more specific about the precise - if symptomatic - meaning of do not. On 01/23/2010 09:08 PM, Zane C.B. wrote: I've run into a odd issue where inbound calls to the SIP client work fine, but outbound from the SIP client do not. The path between the client and the server is as below. N900 SIP client-- OpenVPN -- Asterisk The version of Asterisk in question is 1.6.0.18. Any suggestions? -- Alex Balashov - Principal Evariste Systems LLC Tel : +1 678-954-0670 Direct : +1 678-954-0671 Web : http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users