[Asterisk-Users] Re-Invite between SIP phones
Michael Koehler
koehler at nikotel.com
Tue Jan 20 10:23:31 MST 2004
basically for (re)negotiation on session parameters like:
media codecs,
media IP and PORT
In common it is useful to put a line on hold by setting the media IP to
0.0.0.0
or for a soft redirection of the media stream to another IP and/or PORT.
On the second hand there is a feature called "timer" to check for
aliveness of a session.
Al wrote:
>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 <nkans at lycos.co.uk> wrote:
>
>
>>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 =>
>>username:password at domain.com/extension
>>
>> [extension]
>> redirect=yes
>> redirecturi=sip:12345 at domain1.com
>> redirecturi=sip:56442 at domain2.com
>> ...
>>
>> 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.
>>
>>
>>
>>
>>
>http://bugs.digium.com/bug_view_page.php?bug_id=0000879
>
>
>> 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" <ALow at Prioritytelecom.com>
>>To: <asterisk-users at lists.digium.com>
>>Sent: Tuesday, January 20, 2004 2:00 PM
>>Subject: RE: [Asterisk-Users] Re-Invite between SIP
>>phones
>>
>>
>>
>>
>>>I'd suggest placing a packet sniffer (tcpdump,
>>>
>>>
>>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.
>>
>>
>>>Please provide more information on your setup ...
>>>
>>>-----Original Message-----
>>>From: Al [mailto:albor5000 at yahoo.com]
>>>Sent: Tuesday, January 20, 2004 2:52 PM
>>>To: asterisk-users at lists.digium.com
>>>Subject: RE: [Asterisk-Users] Re-Invite between
>>>
>>>
>>SIP phones
>>
>>
>>>Already did that, but it's not working.
>>>Al
>>>
>>>--- "Low, Adam" <ALow at Prioritytelecom.com> wrote:
>>>
>>>
>>>>canreinvite=yes within sip.conf entities ...
>>>>
>>>>-----Original Message-----
>>>>From: Al [mailto:albor5000 at yahoo.com]
>>>>Sent: Tuesday, January 20, 2004 2:06 PM
>>>>To: asterisk-users at lists.digium.com
>>>>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
>>>>
>>>>
>>avoid
>>
>>
>>>>having the media going through the server?
>>>>
>>>>Tks,
>>>>Al
>>>>
>>>>__________________________________
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>>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
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>>disclose the contents to any other person
>>
>>
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