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</o:shapelayout></xml><![endif]--></head><body bgcolor=white lang=EN-IE link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I found this link which help me solve this problem<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>On reading SIP.CONF it say we can add additional local nets this seems to solve the problem.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>; You may add multiple local networks. A reasonable set of defaults<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>; are:<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:10.0pt;font-family:"Courier New";color:windowtext'>;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Basically I added the following line to my sip.conf <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>localnet=192.168.97.0/255.255.255.0<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>and then I got 2 way audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>This was such an easy fix when you know how!<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I hope it helps someone else.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Regards<o:p></o:p></p><p class=MsoNormal>Albert<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'>From:</span></b><span lang=EN-US style='font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowtext'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Albert Culleton<br><b>Sent:</b> 17 June 2010 15:21<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> [asterisk-users] Asterisk no audio on calls problem.<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span lang=EN-US>Hi there,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I am trying to setup a configuration that requires me to use SIP and asterisk behind a firewall and over a VPN to a remote office and with some local Phones also.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I can’t use IAX to my provider because they don’t offer it and my handsets ( snom 300 ) also don’t support IAX so it’s all SIP.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>The configuration is a follows<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Asterisk PBX 10.202.17.217/24 ------>| 10/100-Switch |-----> Firewall1 pfsense X.Y.Z.250 -------->ITSP Sip Porvider public internet<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>LocalPhones 10.202.17.1-25/24 -_---->| 10/100-Switch |-----> Firewall2 Watchguard ----->ISP internet Connection <-----Firewall3 | remote office | ----Remote User Phone 192.168.97.74/24<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>There is a Lan2Lan VPN tunnel between the Firewall2 and the Remote Office Firewall3<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>I can Ping the remote office phone from the asterisk PBX at all times.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Now I have my Sip.conf setup with externip= X.Y.Z.250<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>[general]<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>port = 5060<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>bindaddr = 0.0.0.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>context = default<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>allowoverlap=no<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>srvlookup = yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>: externip =<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>externip = x.y.z.250<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>localnet=10.202.17.0/255.255.255.0<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>qualify=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>nat=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>register = xxxxxxx:SipServer/xxxxxxxx<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>limitonpeers=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>allowsubscribe=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>notifyringing=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>notifyhold=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>useclientcode=yes<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>canreinvite=no<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I have pfsense setup to forward ports 5060 and RTP ports over UDP back to the internal asterisk server. And a firewall rule to allow this traffic from only my ITSP SipServer.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I can make a call from any phone on the local phones network to the outside world via the SIP proxy with asterisk in the media stream ( canreinvite=no) <o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I can make a call from the remote user phone to a local phone or to any other phone outside the network but I don’t get any audio .<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>If I remove the IP address X.Y.Z.250 from the externip setting then I can call remote phone to local phones fine and get audio perfect, but I can’t make any outbound calls from local to outside world via my ITSP.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Do I need to setup a STUN server to tell the remote Phones that Asterisk is not on the Public address but rather on the LAN address accessible via the VPN?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Or should I put a second Network Adapter in the Asterisk PBX and Setup Iptables on this removing the firewall from the equation ?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I could send all users to the Public Address X.Y.Z.250 but I want to limit by IP address what is allowed in on this and the remote user has a dynamic IP address on their internet connection. So I want to leave this to the last resort.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Has anyone any suggestions?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Thanks<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US>Albert<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p></div></body></html>