[OpenSIPS-Users] Hairpin routing and Loop avoidance

[Steven E. Ames]

Hi. I have an OpenSIPS and Asterisk setup. Incoming calls come from VoIP Carrier to OpenSIPS. OpenSIPS does some dbaliases because sometimes multiple numbers are assigned to same asterisk extension. This all works great.

However, when asterisk makes a call outward for a number that is actually local then I get a Loop and asterisk is unhappy.

Example: I have 2 incoming DIDs (111-222-3333 and 111-222-3334). On OpenSIPS in dbaliases I translated 111-222-3333 to 111-222-3334 and send it to asterisk. All is fine. On asterisk it knows about 3334 but not 3333. So if another extension on asterisk dials 111-222-3333 it gets to OpenSIPS. OpenSIPS does know about 3333 and knows how to handle it. It converts it to 3334 and sends it back to asterisk. Voila. Loop. Now the actual behavior is what I want but I want to modify the SIP INVITE such that asterisk will accept it and not gripe about the loop.

Any pointers?