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asterisk/channels
Mark Michelson 2389278c4a Merged revisions 113928 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines

Merged revisions 113927 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines

We need to set the persistant_route [sic] parameter for the sip_pvt
during the initial INVITE, no matter if we're building the route set from
an INVITE request or response.

(closes issue #12391)
Reported by: benjaminbohlmann
Tested by: benjaminbohlmann

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@113929 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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