[asterisk-bugs] [Asterisk 0013971]: gtalk web no incoming or outgoing calls
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Oct 13 21:59:34 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=13971
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Reported By: adriavidal
Assigned To: phsultan
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Project: Asterisk
Issue ID: 13971
Category: Channels/chan_gtalk
Reproducibility: always
Severity: major
Priority: normal
Status: ready for testing
Asterisk Version: SVN
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!):
Request Review:
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Date Submitted: 2008-11-25 17:35 CST
Last Modified: 2009-10-13 21:59 CDT
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Summary: gtalk web no incoming or outgoing calls
Description:
Calls from or to asterisk done from or to a web enabled gtalk account are
not connected.
Here a put an example of an incoming call from web to Asterisk
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(0112250) jcovert (reporter) - 2009-10-13 21:59
https://issues.asterisk.org/view.php?id=13971#c112250
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I've merged chan_gtalk.c, res_jabber.c, and jabber.h from
team/phsultan/gmail-voice-video with 1.6.1/head. Patch files are
attached.
Observations for reviewers and testers:
1. Autoprune seems like a bad default. Unless you want to explicitly put
all of your buddies into jabber.conf, turning off Autoprune so that the
"central" buddy registry gets loaded each time asterisk starts seems like
the way to go. I did not change this, though.
2. I found the following problems, and fixed them: (a) the ext string I
am seeing from the Google web client is "pmuc-v1 sms-v1 video-v1 voice-v1",
so I'm recognizing it in addition to "pmuc-v1 sms-v1 camera-v1 video-v1
voice-v1" in gtalk_yuck -- without this the Google web clients I've seen
are not jingle-capable. (b) the code in gtalk_is_answered looks for
"session", but the web client is sending "ses:session". This results in
the pointer "tmp" having the value zero when used in the call to
ast_rtp_set_m_type. I have modified the code to search for both "session"
and "ses:session" and to protect asterisk from a seg fault if neither are
found.
There was a call to ast_rtp_lookup_code @ the end of add_codec_to_answer
which I did not remove.
I have only begun testing this. It seems to work, but there are some
issues I've noticed, which seem to be partly Google's fault, and may have
been caused when I had autoprune set on. Specifically, the web client
seemed to think it was a buddy, but during initial registration, Google was
not returning the web client in the roster, and the web client would not
send a new "subscribe". I believe I've worked around this by temporarily
including a "buddy=..." in jabber.conf.
Issue History
Date Modified Username Field Change
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2009-10-13 21:59 jcovert Note Added: 0112250
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