[asterisk-users] canreinvite=yes problem

[BERGANZ François]

Now, I have :

Client 1
---------Asterisk1--Asterisk2
Client 2

I need that sip sign go to Asterisk2
But RTP go to Asterisk1 and no more.

Where have I to insert canreinvite ?

Thank you



-----Message d'origine-----
De : asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] De la part de Eric
"ManxPower" Wieling
Envoyé : mercredi 3 décembre 2008 19:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] canreinvite=yes problem

canreinvite=yes should work as long as 1) there is no NAT involved 
anywhere in the call path, 2) All legs of the call are using the same 
codec, 3) you do not have the t/T/w/W (and maybe a few other) options to 
the Dial line.

Remember the only way you can really tell if a reinvite happens is by 
looking at the RTP audio.  The SIP signaling will not and has never had 
a "reinvite" feature for signaling.

Why did you post the same message at :23, :28, and :35 mins past the 
hour?  If you need immediate support you should contact Digium support 
and pay for a service contract.


BERGANZ François wrote:
> I need to test canreinvite=yes with 2softphones and 1 asterisk.
> I want to have that :
>
http://www.panoramisk.com/wp-content/uploads/2007/05/asterisk-call-flow-outb
> ridge.png
> But I have that http://www.zimagez.com/zimage/canreinvite.php
> Canreinvite=yes work for all phones or just asterisk?...

-- 
Consulting and design services for LAN, WAN, voice and data.  Based near 
Birmingham, AL.  Now accepting clients worldwide. Contact me for Tellabs 
echo canceling systems.  Also see http://www.fnords.org/skillslist.html


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