Hi list! We are testing out SIPREC in our Acme SBCs. The goal is to be "PBX neutral" for a lack of a better word.
We are using Verba as SRS. Basic incoming and outgoing calls work fine. SRC sends SIPREC INVITE to SRS with metadata containing A och B parties. Now, 100% of our customers that want to use call recording of course sometimes transfers the call within the PBX or out to the PSTN. The PBX then sends a re-INVITE or UPDATE to the Acme SBC with a P-Asserted-Identity containing the new participant. When we use port mirror (SPAN) with Verba, they have support for PAI participant change and will start/stop recording based on it. This is not the case with SIPREC. There is no P-Asserted-Identity sent in the SIPREC XML. I've looked at the siprec draft 18 and it mentions P-Asserted-Identity as an additional AOR. Has this specific transfer case without REFER been discussed? Is there any leverage that I can use towards the vendor to have them implement this? (I.E follow the standard). Thanks! _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
