[asterisk-users] OpenVPN + SIP configuration?

[Gilles]

On Thu, 13 Jan 2011 09:43:26 -0500, Bruce B <bruceb444 at gmail.com>
wrote:
>In sip_nat.conf you need to specify 10.8.0.1/24 as your localnet and also
>make sure you have your externip setup as well. Else you will notice one way
>audio or cut off after 30 seconds.

I don't have sip_nat.conf, as I don't use any GUI to configure
Asterisk.

I didn't have to change anything to Asterisk as compared to when
connecting directly.

Since the other extensions live in the same LAN as Asterisk, should I
configure "localnet" just for the remote extension that connects in
through OpenVPN, while leaving 192.168.0.0/24 for the local
extensions?

The only issue I notice, is that Asterisk doesn't tell the other end
when the local end has hung up, so the other end either remains online
or hangs up after 20-30 seconds.
I've tried XLite and ZoIPer, same result. This never happens when not
going through the VPN. Has someone seen this?

Here's the error message:

============
    -- Executing [siemens at internal:1] Dial("SIP/remote-00d22b1c",
"SIP/siemens") in 
new stack
    -- Called siemens
    -- SIP/siemens-00d329ec is ringing
    -- SIP/siemens-00d329ec answered SIP/remote-00d22b1c
    -- Packet2Packet bridging SIP/remote-00d22b1c and
SIP/siemens-00d329ec
  == Spawn extension (internal,siemens, 1) exited non-zero on
'SIP/remote-00d22b1c'

WARNING[82]: chan_sip.c:1948 retrans_pkt: Maximum retries 
exceeded on transmission NWQ2NTRhMzYxZjIzZTBhODY3NTBhYzMxMTk5MTUyYjY. 
for seqno 2 (Critical Response)
============

Thank you.