REINVITE on or off?
Submitted by ja133 on Sun, 04/20/2008 - 7:46pm.
What would be the difference with REINVITE on and off? I just read 2 posts that it will take off a load from the server to pass audio between endpoints only.
Is this better to use?
Will it save or consume bandwidth?
Any reason why you think I should leave it off or turn it on?
Thanks
Joe,
If Asterisk is not in
Sun, 04/20/2008 - 8:05pm
Joe,
If Asterisk is not in the middle of the media stream you loose a bunch of functionality. The ability to use any feature codes while in a call is one of the largest losses.
You also loose OSLEC and any other stream enhancements.
Most importantly if you are using a SIP provider it probably won't work without a proxy. Asterisk is not a SIP proxy nor a Back to Back UA, though some people think of * as a B2BUA.
Since Asterisk supports far more than SIP it is hard to classify * into the IETF RFC 3261
You said in another post you want to get into the blood and guts of Asterisk.
Understanding SIP is a great place to start.
Here is some bed time reading for you:
http://www.ietf.org/rfc/rfc3261.txt
Scott
--
Scott
aka "Skyking"
Will look into that
Sun, 04/20/2008 - 8:10pm
Will look into that link
Thanks for the clarification
I think the only in call
Sun, 04/20/2008 - 8:27pm
I think the only in call feature codes you lose are the following:
In-Call Asterisk Attended Transfer
In-Call Asterisk Blind Transfer
In-Call Asterisk Toggle Call Recording
If those are important IE if the user has an analog phone and/or users need to enable call recording on themselves on the fly, then you should leave canreinvite set to no. Otherwise in order for canreinvite=yes to work, you will need to remove the 't' option from the General Settings>Asterisk Dial command options text box. If you need a specific SIP extension to have these features, then you will need to set this setting in FreePBX: Extensions>extension number>Device Options>dial to SIP/XXXX||tr (where XXXX is your extension number)
If that is wrong, will someone either post a correction or edit my post?
I have been eating,
Sun, 04/20/2008 - 8:59pm
I have been eating, sleeping, and breathing asterisk/freepbx for the past month :-) It's addicting
Before that I dabbled every now and then with VoIP, but nothing major. I outfitted a small shop with a Fonality box about a year and 4 months ago, and kinda made the engineers mad at me for implementing a TFTP server and writing configs for the Aastra phones (I wasn't about to pay umpteen thousand dollars in support costs to get a firmware update for the DST change and to fix the config on a few phones.) Sorry admins, please don't ban me.
Member Since:
2006-11-26