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

elguero reviewboard at asterisk.org
Tue May 15 19:31:26 CDT 2012

This is an automatically generated e-mail. To reply, visit:

Review request for Asterisk Developers.


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.

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.

This addresses bug ASTERISK-19856.


  branches/10/channels/chan_sip.c 366599 

Diff: https://reviewboard.asterisk.org/r/1923/diff


Tested on local machine.  Posted patch to issue tracker and hopefully the reporter will test it out as well.



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120516/7fa9378f/attachment.htm>

More information about the asterisk-dev mailing list