Hello, 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?... Can you help me? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081203/9e1da311/attachment.htm
Someone have a solution for me ? De : asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] De la part de BERGANZ Fran?ois Envoy? : mercredi 3 d?cembre 2008 18:24 ? : asterisk-users at lists.digium.com Objet : [asterisk-users] canreinvite=yes problem Hello, 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?... Can you help me? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081203/1087788e/attachment-0001.htm
On Wed, Dec 03, 2008 at 06:23:32PM +0100, BERGANZ Fran?ois wrote:> Hello, > > 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-outbridge.png > > But I have that http://www.zimagez.com/zimage/canreinvite.php > > > Canreinvite=yes work for all phones or just asterisk?...I believe canreinvite=yes is the default option unless you set it to canreinvite=no I would leave it set to yes unless there is some reason to change it, for example the phone is behind NAT, or transfers etc don't work correctly without it being set to no. If it's still not doing the right thing, then it's worth also checking the nat= option There are also other settings which can cause asterisk to stay in the media path, as BOTH sip devices need canreinvite=yes, otherwise it will stay in the media path. Specifying certain options on the Dial() cmd may also cause it to stay in the media path. Rob
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 Fran?ois 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