I’m looking for a solution in a system with more than a pbx system (i only have access to one of them…):
When I receive a call I have to simulate an hold-unhold action with my asterisk server, basically I only need to transfer RTP flow to a different IP address for a while.
My actual workaround is:
1 ASTERISK SERVER [version 13.8.1] and 1 SIP CLIENT on a windows machine
using a sip client from a different IP registered on asterisk server and to simulate hold/unhold I simply transfer to the SIP client the call and after some seconds re-transfer the channel on server.
in order to reduce the architecture of system, my question Is :
Is it possible to do that using only 1 ASTERISK SERVER with two interface (for example) based on a linux server
- asterisk server on a first IP (ex. 192.168.0.100:5060)
- asterisk SIP client (SIP/75) on a second IP (192.168.0.200:5060) registred on first IP
I’ve tried with different configuration but I receive 482 Loop Detected
[Oct 3 16:12:41] == Using SIP RTP CoS mark 5
[Oct 3 16:12:41] – Called SIP/75
[Oct 3 16:12:41] – Got SIP response 482 “Loop Detected” back from 192.168.0.100:5060
[Oct 3 16:12:41] – SIP/75-00000001 is circuit-busy
[Oct 3 16:12:41] == Everyone is busy/congested at this time (1:0/1/0)