[asterisk-dev] [Code Review] Store route-set from provisional SIP responses so early-dialog requests can be routed properly

[Kevin Fleming]

-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1505/
-----------------------------------------------------------

(Updated Oct. 11, 2011, 10:22 a.m.)


Review request for Asterisk Developers and Olle E Johansson.


Summary (updated)
-------

i have found a problem with a blond transfer and connected_line UPDATE messages. 
When doing a transfer to a ringing channel the UPDATE message will not have any route information attached cause the route header is only parsed on a 200 response and not on a 180.

this small patch parses the contact header and also sets the proper route information even on a 180 respsonse so when an UPDATE messages is sent out, the route header is attached.

maybe doing the same for a 183 response would also be a good idea.


Diffs
-----

  team/schmidts/unleash-the-beast/channels/chan_sip.c 340107 

Diff: https://reviewboard.asterisk.org/r/1505/diff


Testing
-------

asterisk is behind a kamailio sip proxy which drops in-dialog messages with no prober route header. after parsing the route header of a 180 reponse the proxy forwards the information to the right end point.

normal transfer, blind transfer, ringing ... works fine.

CONNECTED_LINE function after the transfer also works like expected.


Thanks,

schmidts

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20111011/72b168a8/attachment-0001.htm>