[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

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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