I can't use {SIPDOMAIN} because the {SIPDOMAIN} variable is actually the IP address of callers phone as it appears in the location table.
On a side note I was able to not use P-Asserted-Identity. because of a different issue I learned about the uac_replace_to() function. I was able to place the real domain in the TO header. Now with Asterisk I do the following exten => _VMS_.,1,Ringing exten => _VMS_.,n,Wait(1) exten => _VMS_.,n,Answer exten => _VMS_.,n,Wait(1) exten => _VMS_.,n,Set(dm=${SIP_HEADER(TO):16}) exten => _VMS_.,n,Set(dm=${CUT(dm,>,1)}) exten => _VMS_.,n,Voicemail(${EXTEN:4...@${dm},u) exten => _VMS_.,n,Hangup The offsets I use above assumes the username of the extension being called is a 10 digit users NPANXXXXXX Thanks for the reply. The uac_replace_to() fixed multiple issues. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Asterisk-Integration-Manipulate-Asterisk-Contexts-tp5711054p5717739.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users