SIPv6 running dual-stack Hi everyone,
I have a 3825 acting as SBC on my WAN edge and a 2821 running CM in the backend, the phones are registered to the 2821 using skinny. SIP over IPv4 is up and running since a while.
I've enabled IPv6 on the network a little while ago, active directory and DNS is running 100% v6.
For PSTN we use a SIP carrier that only supports v4 for now, however, I tought that getting SIPv6 ready on my end would be a great idea for the future.
So basically, I'm trying to receive calls from the PSTN in v4 on 1 call leg to the SBC, and a v6 call leg between the SBC and CME.
So, I've enabled dual-stack in the sip-ua, changed my session target to the v6 address and...........the phone rings but there is no audio. Transfers to VM fails as well but i'll address this seperately.
I've read on SIPv6 and saw some mention about MTP that would translate the v5 audio to v6. I tried a few things without much results.
I'm a routing & switching net admin and I have a pretty good expertise on SIP but I have no clue where to start with on this IPv6 audio problem. I don't know where to start because the SBC and the CME are only 2 hops away, without any security appliance in between, so I have no clue why I'm not even hearing the "ringing"
The SBC SIP process is not bound to any interfaces, the CME is bounded to is Loopback0.
Any links to some usefull documentation or any hint would be greatly appreciated.