[asterisk-bugs] [Asterisk 0010291]: Skinny to skinny calls one-way audio
noreply at bugs.digium.com
noreply at bugs.digium.com
Thu Jul 26 18:26:52 CDT 2007
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=10291
======================================================================
Reported By: sbisker
Assigned To: qwell
======================================================================
Project: Asterisk
Issue ID: 10291
Category: Channels/chan_skinny
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!): 76784
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 07-24-2007 11:13 CDT
Last Modified: 07-26-2007 18:26 CDT
======================================================================
Summary: Skinny to skinny calls one-way audio
Description:
Version 76621 fixed the crashing of asterisk due to soft-key problems, but
introduced another problem.
7921 -> 7921 phones the remote party could not hear any audio. In version
76784, the 7921 that initiates the call hears no audio. If a 7920 phone
calls a 7921/7920 phone, then both parties hear audio.
======================================================================
----------------------------------------------------------------------
DEA - 07-26-07 18:26
----------------------------------------------------------------------
The 7920 and I am guessing the 7921 do send the call reference, but the
key is for an established connection. Starting from the onhook state,
there
is no active sub, so the phone send 0.
The back2backcalls log show the phones do send the call reference.
I continue to like the approach in the transfer patch and have already
thought about how it can help with additional feaures. That said, if the
phone provides us with callreference information, we should use it.
If it does not provide us that detail, then we can fall back to the
channels
view of the world. Do you know if the 30VIP phones can support more than
one active call? Having the callreference is going to be key to
supporting
multiple calls per line.
I also agree that the code in trunk and 1.4 has problems with handling
subs.
I do not see where we prune subs from the lines list, meaning that we are
wasting memory. One of my tries at the transfer patch included an
attempt
to remove a sub at hangup, and this is not a unique requirement for
transfer.
It impacts phones that want to have multiple calls, with one or more on
hold.
Wedhorn, I think we've swapped emails before. Can you send me the CCM
traces of the 30VIP in action?
Issue History
Date Modified Username Field Change
======================================================================
07-26-07 18:26 DEA Note Added: 0067954
======================================================================
More information about the asterisk-bugs
mailing list