Thanks for that Carlos. I am playing with that right now. What do you suggest localnet should say?
Server A = OpenVPN Server: localnet=127.0.01 localnet=192.168.100.0/255.255.255.0 Where 192.168.100.0 is the DHCPd subnet of Server B (the openvpn client) Server A doesn't have any localnet other than the loop back and then a Vnet to internet (public ip address). Thanks, Bruce On Wed, Sep 22, 2010 at 11:36 AM, Carlos Chavez <[email protected]>wrote: > Do you have a localnet statement in your sip.conf? That and using > nat=no will make sure Asterisk does not replace the IP address in the > Invite. > > On Wed, 2010-09-22 at 01:27 -0400, bruce bruce wrote: > > 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 > > 192.168.100.0/24 address 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, > > -- > > _____________________________________________________________________ > > -- 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 > -- > Carlos Chavez > Director de Tecnología > Telecomunicaciones Abiertas de México S.A. de C.V. > Tel: +52-55-91169161 Ext 2001 > > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
