[asterisk-dev] [Code Review] Fix for SIP peer's allowtransfer setting not being used

[Mark Michelson]

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I suspect this will only work for incoming calls for the peer. If a call is placed to the peer, then the allowtransfer setting for that peer still will not be honored.

There are two places where peer settings are typically copied to a sip_pvt.

1. For outbound calls, check create_addr_from_peer().
2. For inbound calls, check check_peer_ok().

Copy the value in those two functions and things should be good to go.

- Mark


On May 15, 2012, 7:31 p.m., elguero wrote:
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> (Updated May 15, 2012, 7:31 p.m.)
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> Review request for Asterisk Developers.
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> Summary
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> When setting the global setting in sip.conf for allowtransfer to "no" and then setting the peer's allowtransfer setting to "yes", call transfers are being denied.  This would appear to be caused by the dialog not being set to use the peer's allowtransfer setting.  The allowtransfer setting is set to the default global setting at the time of creation but never updated to reflect the related peer's setting.
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> This patch attempts to fix this by setting the dialog's allowtransfer to that of the peer's when there is a related peer found.
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> This addresses bug ASTERISK-19856.
>     https://issues.asterisk.org/jira/browse/ASTERISK-19856
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> Diffs
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>   branches/10/channels/chan_sip.c 366599 
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> Diff: https://reviewboard.asterisk.org/r/1923/diff
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> Testing
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> Tested on local machine.  Posted patch to issue tracker and hopefully the reporter will test it out as well.
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> Thanks,
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> elguero
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>

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