Asterisk bounty symmetic SIP invites

Bounty Paid: 12/8/2004 to Theo Zourzouvillys

Originally documented in http://bugs.digium.com/bug_view_page.php?bug_id=0002358.

Currently if we have Asterisk SIP channel driver binding to all interfaces,
and eth0 has many subnets attached to it (a primary 10.1.200.1, and then
alias interfaces eth0:1 with 10.1.201.1, eth0:2 with 10.1.202.1, eth0:3
with 10.1.203.1..

If an INVITE is sent to Asterisk on 10.1.202.1 (eth0:2) the response is
always returned to 10.1.200.1. We need it to come back to 10.1.202.1.

Spoke to Mark Spencer about this, and he agreed do to the nature of UDP
there would have to be code written to support this. I inquired with
Digium to do it, but haven't heard back from their consulting folks, so
I'm asking the greater community as a whole.

Our company is willing to pay a $500 bounty (Paypal) to code sent to us,
that we'll approve and agree works for our load test scenarios we develop
in house and then send/submit to the Asterisk folks as open source.

It should be a patch/diff on the -HEAD version of chan_sip.c (or any other
code that might need to be changed).
Bounty Paid: 12/8/2004 to Theo Zourzouvillys

Originally documented in http://bugs.digium.com/bug_view_page.php?bug_id=0002358.

Currently if we have Asterisk SIP channel driver binding to all interfaces,
and eth0 has many subnets attached to it (a primary 10.1.200.1, and then
alias interfaces eth0:1 with 10.1.201.1, eth0:2 with 10.1.202.1, eth0:3
with 10.1.203.1..

If an INVITE is sent to Asterisk on 10.1.202.1 (eth0:2) the response is
always returned to 10.1.200.1. We need it to come back to 10.1.202.1.

Spoke to Mark Spencer about this, and he agreed do to the nature of UDP
there would have to be code written to support this. I inquired with
Digium to do it, but haven't heard back from their consulting folks, so
I'm asking the greater community as a whole.

Our company is willing to pay a $500 bounty (Paypal) to code sent to us,
that we'll approve and agree works for our load test scenarios we develop
in house and then send/submit to the Asterisk folks as open source.

It should be a patch/diff on the -HEAD version of chan_sip.c (or any other
code that might need to be changed).
Created by: InetNomad, Last modification: Wed 08 of Dec, 2004 (21:40 UTC)
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