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

[Kevin Fleming]

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Ship it!


Nice work finding and fixing this one.

- Kevin


On Oct. 11, 2011, 10:22 a.m., schmidts wrote:
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> (Updated Oct. 11, 2011, 10:22 a.m.)
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> Review request for Asterisk Developers and Olle E Johansson.
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> Summary
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> 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.
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> 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.
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> maybe doing the same for a 183 response would also be a good idea.
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> Diffs
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>   team/schmidts/unleash-the-beast/channels/chan_sip.c 340107 
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> Diff: https://reviewboard.asterisk.org/r/1505/diff
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> Testing
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> 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.
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> normal transfer, blind transfer, ringing ... works fine.
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> CONNECTED_LINE function after the transfer also works like expected.
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> Thanks,
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> schmidts
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