[Asterisk-Users] Remote SIP client HACK JOB
Ryan Courtnage
ryan at voxbox.ca
Wed Jun 30 09:31:48 MST 2004
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
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