Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <6369cb70bfd88942b9705ac1e639a33821fd957...@dc-us1mbex4.global.avaya.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45407> Message-ID: <b15f.4bd1d...@forum.sipfoundry.org>
Hi Dave, I have managed to get a wireshark trace of the divert to voicemail. The key bit of the decode are: From: "6667912" <sip:mailto:6667...@10.203.105.50>;tag=d19c5205-82bd-44fc-88c4-bf5d3c52feb5-37461274 Diversion: "6667912" <sip:mailto:6667...@10.203.105.50>;reason=unconditional;privacy=off;screen=yes To: <sip:mailto:71...@10.203.104.128> The actual call flow is 71901 > 6667912 > forwards to vmail pilot number 71900. I noticed that the 'From' address is always the same as the diversion, is this okay? So, the original calling party id is lost. When the AA answers the call, it appears that it thinks the call is from the mailbox owner and prompts for PIN. Can you confirm whether the From address has to be be the original calling party. In addition I noticed that the format of the diversion header was slightly different in that there is no 'tel:' but just the number - does this cause any issues? Thanks for your help Abdul Thanks Abdul _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
