Hairpinned call (SIP to SIP same provider)

[gpworld]

I am working on a client site and trying to get CUBE to hairpin an inbound call from a SIP provider back out to an answering service in the event the CUCM is down.  ISP SIP trunk <--> FW(NAT) <--> CUBE <--> CUCM.  The routing works (call signaling) but there is no audio in the connected call.  I am using a secondary dial-peer to trigger the re-routing.  Any suggestions as to why no audio?  Have tried several things.

[R0g22]

Any logs from the CUBE that we can look at ?

[Marcelo Morais]

Hi @gpworld,

 what is probably going on is that the SIP packet received by the ISP contains the "Answering Service" IP Addr (CUBE's local IP Addr) not the CUBE's "NAT" IP Addr, in other words, the packet is going to the wrong path.

 If this is the case you need to configure a voice class sip-profile to modify the SDP Header - from the "Answering  Service" IP Addr to the CUBE's "NAT" IP Addr.

Hope this helps,

 Marcelo Morais