I couldn't be happier with the simplicity of this - but it's a hack! Hi all,
I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: - Asterisk's firewall forwards 5060 udp and 10000-20000 udp to * - The SIP client's firewall forwards 5060 udp and 10000-20000 udp to the SIP client - sip.conf contains NAT=YES for this particular client - The SIP client has no special settings, just the external IP of Asterisk's firewall for the SIP Server. I couldn't be happier with the simplicity of this ... however, here is the HACK JOB I need to perform to get the external SIP client's audio to work: When I first start up Asterisk, I need the following In SIP.CONF's [genera] section: - bindaddr = 0.0.0.0 This allows all my internal office phones to work, and also allows me to dial to/from my external client. However, the external client will hear/send no audio. To allow the external client to hear/send audio, I have to change sip.conf ... - bindaddr = <EXTERNAL IP> ... followed by issuing a "RELOAD" at the * CLI. It's a total hack, cause if I try to START/RESTART Asterisk with "bindaddr=<EXTERNAL IP>", neither the internal or external clients will work, and I'll just see a bunch of this on the console: Jun 30 15:51:28 WARNING[-1275102288]: chan_sip.c:590 __sip_xmit: sip_xmit of 0x80f680c (len 465) to 66.18.203.117 returned -1: Bad file descriptor My question is ... "Is there a better way to do this, without the use of STUN or proxy servers?" Thanks -- .................................. Ryan Courtnage Coalescent Systems Inc 403.244.8089 www.voxbox.ca _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
