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basically for (re)negotiation on session parameters like:<br>
<br>
media codecs,<br>
media IP and PORT<br>
<br>
In common it is useful to put a line on hold by setting the media IP to
0.0.0.0<br>
or for a soft redirection of the media stream to another IP and/or PORT.<br>
<br>
On the second hand there is a feature called "timer" to check for
aliveness of a session. <br>
<br>
<br>
<br>
<br>
<br>
Al wrote:<br>
<blockquote type="cite"
cite="mid20040120160045.4125.qmail@web40402.mail.yahoo.com">
<pre wrap="">
What I would like to understand is in what situations
reINVITEs are issued.
Anyway, I got the following messages when trying to
apply your patch.
patching file chan_sip.c
Hunk #1 succeeded at 160 with fuzz 1.
Hunk #2 succeeded at 365 with fuzz 1.
Hunk #4 FAILED at 2253.
Hunk #5 FAILED at 2291.
Hunk #6 FAILED at 5019.
Hunk #7 FAILED at 5168.
Hunk #8 succeeded at 5911 with fuzz 1.
Hunk #9 FAILED at 6245.
Hunk #10 FAILED at 6397.
patch unexpectedly ends in middle of line
Hunk #11 FAILED at 6683.
7 out of 11 hunks FAILED -- saving rejects to file
chan_sip.c.rej
Al
--- Kannaiyan Natesan <a class="moz-txt-link-rfc2396E" href="mailto:nkans@lycos.co.uk"><nkans@lycos.co.uk></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi,
I think canreinvite=yes won't work in most of the
situations.
I have implemented Redirect SIP 300 Message to
redirect to the SIP
address you speficy in the sip.conf.
Where you can have ,
register =>
<a class="moz-txt-link-abbreviated" href="mailto:username:password@domain.com/extension">username:password@domain.com/extension</a>
[extension]
redirect=yes
<a class="moz-txt-link-abbreviated" href="mailto:redirecturi=sip:12345@domain1.com">redirecturi=sip:12345@domain1.com</a>
<a class="moz-txt-link-abbreviated" href="mailto:redirecturi=sip:56442@domain2.com">redirecturi=sip:56442@domain2.com</a>
...
will make to redirect to all the URI's yu
specify in the sip.conf. I'm
also working on this so that it can get the
redirections from the database
rather than reloading asterisk all the time when you
modify the redirection
uri.
You can check through that.
</pre>
</blockquote>
<pre wrap=""><!----><a class="moz-txt-link-freetext" href="http://bugs.digium.com/bug_view_page.php?bug_id=0000879">http://bugs.digium.com/bug_view_page.php?bug_id=0000879</a>
</pre>
<blockquote type="cite">
<pre wrap=""> Message transmission is alright, but for some
reason it is not working.
Can you test with yours and let me know where is the
problem, I will modify
the code once you get the clue where is the problem
on it. If successfully
please send me the sip debug message and I will just
make sure it works for
all.
Kannaiyan
----- Original Message -----
From: "Low, Adam" <a class="moz-txt-link-rfc2396E" href="mailto:ALow@Prioritytelecom.com"><ALow@Prioritytelecom.com></a>
To: <a class="moz-txt-link-rfc2396E" href="mailto:asterisk-users@lists.digium.com"><asterisk-users@lists.digium.com></a>
Sent: Tuesday, January 20, 2004 2:00 PM
Subject: RE: [Asterisk-Users] Re-Invite between SIP
phones
</pre>
<blockquote type="cite">
<pre wrap="">I'd suggest placing a packet sniffer (tcpdump,
</pre>
</blockquote>
<pre wrap="">etherreal) and see whats
happening because it works great for me and always
has but I guess it also
requires support on the end-points and possibly
(assuming non-cisco enviro)
there maybe an option that needs to be configured on
your phones/gateways.
</pre>
<blockquote type="cite">
<pre wrap="">Please provide more information on your setup ...
-----Original Message-----
From: Al [<a class="moz-txt-link-freetext" href="mailto:albor5000@yahoo.com">mailto:albor5000@yahoo.com</a>]
Sent: Tuesday, January 20, 2004 2:52 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: RE: [Asterisk-Users] Re-Invite between
</pre>
</blockquote>
<pre wrap="">SIP phones
</pre>
<blockquote type="cite">
<pre wrap="">
Already did that, but it's not working.
Al
--- "Low, Adam" <a class="moz-txt-link-rfc2396E" href="mailto:ALow@Prioritytelecom.com"><ALow@Prioritytelecom.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">canreinvite=yes within sip.conf entities ...
-----Original Message-----
From: Al [<a class="moz-txt-link-freetext" href="mailto:albor5000@yahoo.com">mailto:albor5000@yahoo.com</a>]
Sent: Tuesday, January 20, 2004 2:06 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: [Asterisk-Users] Re-Invite between SIP
phones
Anybody knows what do I need to tell Asterisk
to issue a re-INVITE between two SIP phone to
</pre>
</blockquote>
</blockquote>
<pre wrap="">avoid
</pre>
<blockquote type="cite">
<blockquote type="cite">
<pre wrap="">having the media going through the server?
Tks,
Al
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<pre wrap="">
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This message and any attachment are confidential
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<pre wrap="">If you are not the intended recipient, please
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********* DISCLAIMER *********
This message and any attachment are confidential
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<pre wrap="">and may be privileged or
otherwise protected from disclosure and may include
proprietary information.
If you are not the intended recipient, please
telephone or email the sender
and delete this message and any attachment from your
system. If you are not
the intended recipient you must not copy this
message or attachment or
disclose the contents to any other person
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