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(a)sip.com for sip-id
5551212(a)sip.com.
When 1000(a)sip.com is dialed by another user in the
domain (I am using "use_domain") the call is properly
routed to: 5551212(a)sip.com.
However, if 5551212(a)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(a)sip.com
Needs to be:
665551212(a)sip.com
Thoughts?
F
--- Bogdan-Andrei Iancu <bogdan(a)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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