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Summary:ASTERISK-16026: sip_pvt does not have association with sip_peer for INVITEs
Reporter:Philip Prindeville (pprindeville)Labels:
Date Opened:2010-04-28 12:42:49Date Closed:2010-07-10 09:32:49
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Description:Registrations, Options, etc. point back to their associated peer but oddly calls do not.  This change makes the association more consistent for all message types.


****** ADDITIONAL INFORMATION ******

Tested on 1.6.2.6 and confirmed to work.

Also note that this fix is precursory to:

https://reviewboard.asterisk.org/r/565/
Comments:By: Russell Bryant (russell) 2010-05-11 10:49:12

Is this still relevant since review 565 is closed?

By: Philip Prindeville (pprindeville) 2010-05-11 11:06:01

Yes, this is still relevant.  It remains useful to be able to see what peering information was used to bring up a call.

We're using this here in production with Astlinux and it works reliably.

By: Russell Bryant (russell) 2010-07-10 09:32:49

patch has been committed already