Good point. I need to clarify...
The alias is working to route to the user however if the timer expires, the route fails or if there is a reject, the failure routes are applying their logic to the alias-id not the sip-id.
What I need to do is apply the failure route logic to the sip-id so we can send the attempt to voicemail.
Let say I have an alias of 1000@sip.com for sip-id 5551212@sip.com.
When 1000@sip.com is dialed by another user in the domain (I am using "use_domain") the call is properly routed to: 5551212@sip.com.
However, if 5551212@sip.com does not answer, timer expires, or a reject, then the failure logic is be applied to the alias.
Failure route says:
prefix("66"); append_branch(); append_urihf("CC-Diversion: ", "\r\n"); t_relay();
Result of the failure logic:
661000@sip.com
Needs to be:
665551212@sip.com
Thoughts?
F
--- Bogdan-Andrei Iancu bogdan@voice-system.ro wrote:
Hi,
not sure I understand - you dial an alias and via db_aliases you replace the alias with the sip-id. So you have the sip-id .. where is the problem?
regards, bogdan
Frogger wrote:
I am using alias_db to store alias's for users.
However when an alias is dialed I need to also be
able
to restore the alias owners original sip-id.
This is necessary to forward the call to voicemail using sip-id.
Any guidance on getting the sip-id out of the
alias_db
and replacing the alias-id prior to forwarding to vmail?
Thanks in advance!
F
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