[OpenSIPS-Users] Asterisk Integration - Manipulate Asterisk Contexts
osiris123d
duane.larson at gmail.com
Mon Nov 8 18:25:55 CET 2010
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.
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