[asterisk-users] canreinvite=yes problem

[Eric "ManxPower" Wieling]

Reinvites will happen by default.  Post your sip.conf [general] and the 
peers in sip.conf masking only the passwords.  Also paste the part of 
extensions.conf that you use to Dial.

BERGANZ François wrote:
> 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?...
> 

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