----- Original Message ----- > From: "Administrator TOOTAI" <ad...@tootai.net> > To: asterisk-users@lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a écrit : > > ----- Original Message ----- > >> From: "Administrator TOOTAI" <ad...@tootai.net> > >> To: asterisk-users@lists.digium.com > >> Sent: Thursday, April 30, 2015 4:43:33 PM > >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call > >> In > >> > >>> I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > >>> internal phones are located on the 10.10.32.0/21 LAN subnet. I have many > >>> internal SIP phones, which appear to be working correctly. I have a few > >>> external phones (Yealink SIP-T32G or other Yealink model) on > >>> 192.168.32.0/24 which have an OpenVPN client configured on them that > >>> connects back to the LAN network through a pfSense gateway with OpenVPN > >>> configured on it. > >> > >> I faced problems with pfsense -no VPN involved- and finally installed > >> siproxd on it. Also set the firewall mode to conservative. > > > > Daniel, > > > > Thanks for the information. Do you have an example or documentation on the > > siproxd configuration that you used? > > No, just follow the basis of the parameters given by the package. If I > remember, SIP use the proxy siproxd and RTP is direct. >
Looking into it further, in my case it does not appear to be a NATing issue, since running OpenVPN from pfSense means there's no NATing occurring between the clients or between the clients and the asterisk server. Although I was unable to reproduce the problems, I did notice some packet loss and jitter in "sip show channelstats", here is a sample: Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 192.168.32.26 446613544@1 00:03:03 0000000094 0000004238 (97.83%) 0.0000 0000000000 0000000244 ( 0.00%) 0.0000 192.168.32.38 5b2ebdc92fd 00:03:03 0000000059 0000000001 ( 1.67%) 0.0000 0000000000 0000000091 ( 0.00%) 0.0028 I was unable to find documentation each of these columns, but the high percentage of loss for received packets for 192.168.32.26 seems suspicious. Do these statistics indicate a problem? Thanks, Andrew > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users