[Asterisk-Users] CODECs and sip.conf and voice quality

[Mike Meyer]

Robert,

	RE: The reinvite=yes option. 

	My answer is that I am not sure, but I think you are right. Based on
the descriptions I have found ...

a)http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly
b)http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
 
it sounds like the Asterisk is still in the signalling path with it set
to yes. From previous investigation to support call parking,

c)http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20parking

documentation indicates it is to be set to yes for that to work.
Probably for the same reason though, so that call supervision can detect
the # to transfer and must be kept in the media path.

In my case, I am probably getting away with it set to yes since the dial
has tT option set and I am connected a call between a SIP phone and a
TDM card so it won't reinvite in either case.

I can't remember the reason that I had now for setting it to yes. I may
have just gotten totally confused. Easy to do. All these options and
dependencies keep my head spinning!

In final; I tested ilbc with canreinvite=no just for kicks. Transfer
still does not work with the #. Had to use the transfer button on the GS
phone.

Thanks again for your comment,
Mike



>Date: Tue, 28 Sep 2004 13:24:53 -0400
>From: "Robert Jackson" <RobertJ at promedicalinc.com>
>Subject: RE: [Asterisk-Users] CODECs and sip.conf and voice quality
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>        <asterisk-users at lists.digium.com>
>Message-ID:
>       
><2BE8436A70E4AF4C9663B9E98CD52B763880C8 at promed_2.promedicalinc.com>
>Content-Type: text/plain;       charset="US-ASCII"
>
>> -----Original Message-----
>> From: Mike Meyer [mailto:mjmeyer at gendesign.com] 
>> Sent: Tuesday, September 28, 2004 1:07 PM
>> To: Asterisk Users Group
>> Subject: [Asterisk-Users] CODECs and sip.conf and voice quality
>> 
>> 
>> Another Caveat:
>> Transfer does not work using the # key with the ILBC CODEC on 
>> the GS phones. I can transfer only with the transfer button. 
>> I have asterisk in the loop doing call supervision since I 
>> have the tT option set in the dial command and 
>> canreinvite=yes for the SIP phones. Anyone else have this problem?
>> 
>
>Shouldn't canreinvite be set to no to keep the phones from 
>reinviting? I agree that with the tT flags * should still be
>in the path, but I was just curious.