Asterisk with Failover - Pacemaker

Asterisk Asterisk General
1

hello,
I configured the two asterisk node using pacemaker and has created the floating IP.

I use floating IP as sip registration IP in all phones. Since the floating IP can move to node2 in case of node1 failure or problem and call service continue from node2.

Floating IP: 172.29.1.82
eth0 IP: 172.29.1.81

I register sip ext 100 to 172.29.1.82
but but when I send SIP invites from 100. it appear as 100@172.29.1.81 which is eth0 IP not floating IP to whom the 100 register. In such case I can make calls but there is one-way audio.

I am thinking to use SIP Outbound proxy as floating iP but not sure if it work.

chan_sip.c:25864 in handle_request_do: --- (14 headers 24 lines) --- Jan 1 11:32:10 G100-59-c4-5f asterisk[18926]: VERBOSE[18992]: chan_sip.c:16098 in check_via: Sending to 172.29.1.81:5060 (NAT) Jan 1 11:32:10 G100-59-c4-5f asterisk[18926]: VERBOSE[18992]: chan_sip.c:22929 in handle_request_invite: Using INVITE request as basis reque st - 5aacb53e3de8654e75721567415b625a@172.29.1.81:5060

2

The contact address will be derived by temporarily binding a socket for the destination address and using the resulting local address. If there are two equally good interfaces, it may not be the interface address used in the request.

By default, asterisk ignores the request URI’s domain, and the destination IP address, so I suspect that configuring the phone to use the floating address as proxy should work.

This is not a discussion topic, so should not have been on this forum.

3

SOLVED!
Hi David,
Thank you for the reply. I actually forgot to add ‘canreinvite=no’ parameter to SIP ext 100.
This solved the problem.
I would like to request to have this post on forum as it may be useful for someone else.