[asterisk-dev] [Code Review] Fix SIP connected line updates.
rmudgett
reviewboard at asterisk.org
Tue Apr 26 17:49:41 CDT 2011
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1199/
-----------------------------------------------------------
Review request for Asterisk Developers and Mark Michelson.
Summary
-------
This patch fixes a couple connected line update problems:
1) The connected line needs to be updated when the initial INVITE is sent
if there is a peer callerid configured. Previously, the connected line
information did not get reported until the call was connected so SIP could
not report connected line information in ringing or progress messages.
2) The connected line should not be updated on initial connect if there is
no connected line information. Previously, all it did was wipe out any
default preset CONNECTEDLINE information set by the dialplan with empty
strings.
This addresses bug 18367.
https://issues.asterisk.org/view.php?id=18367
Diffs
-----
/branches/1.8/channels/chan_sip.c 315663
Diff: https://reviewboard.asterisk.org/r/1199/diff
Testing
-------
Both issues now work as intended.
Thanks,
rmudgett
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20110426/237cdbe4/attachment.htm>
More information about the asterisk-dev
mailing list