I am having an interesting problem with the Pirelli LP-D10 phone and OpenVPN.
First - the Pirelli works fine when I register with Asterisk (both 1.4 and 1.8) outside of OpenVPN.
A bit about my OpenVPN network: I have a tunneled (not bridged) network with a variety of clients connected to my OpenVPN server. Asterisk is running on one of the clients (10.8.0.6) and my Pirelli is connected wirelessly to another client (192.168.5.1) which is an Asus WL-520gu running Tomato VPN. The Pirelli gets a DHCP address (192.168.5.133) from the ASUS router. The router does not have a route to the internet for clients, it can only pass packets onto the VPN.
I've successfully used this network (currently a server and 9 clients) with a variety of different endpoints, connected both wirelessly and wired. I'm confident that the problem is not with the OpenVPN configuration.
My problem is that when I make a call either to or from the Pirelli when it is on the OpenVPN network, the audio drops out completely a few seconds after the call begins. As noted above, this does not happen if I register the Pirelli directly over the internet and not through the OpenVPN network. There's nothing unusual in my Asterisk logs.
I'm not sure how to troubleshoot this problem. I've collected a tcpdump of the SIP packets between the Pirelli and Asterisk that can be viewed with wireshark. This tcpdump has the registration and four phone calls in it - the first two calls are from the Pirelli (extension 701) to another extension (710) and the last two from extension 710 to the Pirelli. In each case, the audio drops after just a few seconds. However, the call doesn't hang up until I end the call by hanging up the phone (same behavior hanging up with either phone).
Its almost like the Pirelli tries to REINVITE the call to send the audio directly, rather than proxying through Asterisk, but I have canreinvite=no on both extensions and there are no REINVITE requests in the tcpdump.
If somebody who understands the SIP protocol better than I could take a look and make a suggestion, I'd be much obliged. I'd be happy to provide more information if that would be helpful.
Thanks.
Dave
First - the Pirelli works fine when I register with Asterisk (both 1.4 and 1.8) outside of OpenVPN.
A bit about my OpenVPN network: I have a tunneled (not bridged) network with a variety of clients connected to my OpenVPN server. Asterisk is running on one of the clients (10.8.0.6) and my Pirelli is connected wirelessly to another client (192.168.5.1) which is an Asus WL-520gu running Tomato VPN. The Pirelli gets a DHCP address (192.168.5.133) from the ASUS router. The router does not have a route to the internet for clients, it can only pass packets onto the VPN.
I've successfully used this network (currently a server and 9 clients) with a variety of different endpoints, connected both wirelessly and wired. I'm confident that the problem is not with the OpenVPN configuration.
My problem is that when I make a call either to or from the Pirelli when it is on the OpenVPN network, the audio drops out completely a few seconds after the call begins. As noted above, this does not happen if I register the Pirelli directly over the internet and not through the OpenVPN network. There's nothing unusual in my Asterisk logs.
I'm not sure how to troubleshoot this problem. I've collected a tcpdump of the SIP packets between the Pirelli and Asterisk that can be viewed with wireshark. This tcpdump has the registration and four phone calls in it - the first two calls are from the Pirelli (extension 701) to another extension (710) and the last two from extension 710 to the Pirelli. In each case, the audio drops after just a few seconds. However, the call doesn't hang up until I end the call by hanging up the phone (same behavior hanging up with either phone).
Its almost like the Pirelli tries to REINVITE the call to send the audio directly, rather than proxying through Asterisk, but I have canreinvite=no on both extensions and there are no REINVITE requests in the tcpdump.
If somebody who understands the SIP protocol better than I could take a look and make a suggestion, I'd be much obliged. I'd be happy to provide more information if that would be helpful.
Thanks.
Dave
Phone System 