Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs.
So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP debug I can see that asterisk uses it's external public IP address to communicate to endpoints that are known to it as the 192.168.100.0/24 endpoints and the endpoints identify themselves with the OpenVPN tunnel IP address scheme in one part of the sip handshake. How can this be fixed? After all, with the OpenVPN this should all look like an internal network to Asterisk. I have added my comments followed by # to lines below that are problematic. <--- SIP read from UDP:192.168.100.5:5060 ---> #This line is good as it uses the local DHCP supplied network address scheme INVITE sip:[email protected]:5060 SIP/2.0 #This line is BAD. Why are we inviting Ext. 203 with it's OpenVPN IP while it's on the same network of 192.168.50.0/24 as 202? Via: SIP/2.0/UDP 192.168.100.5:5060;branch=z9hG4bK695f8c1cfc7cdee96.1c65dc2eb25a46fc6 Max-Forwards: 70 From: "SIP Phone - Ext. 202" <sip:[email protected]:5060>;tag=6d6f8c4226 #BAD line again. Should be SIP:[email protected] <sip%[email protected]> To: "203" <sip:[email protected]:5060> #Bad again.... Call-ID: 43af67a634e06e75 CSeq: 32058 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "SIP Phone - Ext. 202" <sip:[email protected]:5060 ;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D25B72F>" Supported: gruu, path, timer, 100rel, replaces User-Agent: Aastra 55i/2.5.2.1500 Content-Type: application/sdp Content-Length: 594 Basically the phones should only send with FROM their local 192.168.100.0/24address and Asterisk should only send ANSWER and ACK back to 192.168.100.0/24 rather than sending it to 172.16.0.0/24 (which is the openvpn client ip). Once above is fixed, I think all the audio and call cut will go away. I hate to use a sip proxy in this situation since I already have an openvpn connection. Any feed back is appreciated. Thanks,
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