What does your dial command look like? If you have Tt, wW, or hH, then asterisk will always stay in the path.
On 12/8/05, George Pajari <[EMAIL PROTECTED]> wrote: > We are trying to use Asterisk to set up a call between two SIP devices > and then step out of the path. > > - all systems have public IP addresses (no firewalls, no NAT). > - sip.conf has "canreinvite=yes" for both devices > - ulaw is the only permitted codec so we do not have transcoding issues > (and a "sip show channels" confirms both legs at ulaw) > > yet a SIP trace shows that Asterisk does even try to issue a reinvite. > > What else should we look at to see where things are going wrong? > > -- > George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) > Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) > www.netvoice.ca www.ip-centrex.ca > www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
