Hi list, i have the next problem: I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) ) the SIP/111 is a GrandStream ATA the SIP/112 is a Polycom 301 the ZAP/35 is a Analogic Phone. The SIP/112 hints works great. But the other 2 no. The ZAP/35 is say is always in USE and as you see en the next console output is not in use. any Idea???? asterisk*CLI> -= Registered Asterisk Dial Plan Hints =- 111 : SIP/111 State:Idle Watchers 4 102 : ZAP/35 State:InUse Watchers 5 112 : SIP/112 State:InUse Watchers 2 ---------------- - 3 hints registered asterisk*CLI> show cha channel channels channeltypes asterisk*CLI> show channels Channel Location State Application(Data) Zap/34-1 s@incel:1 Up Bridged Call(SIP/112-1f3d) SIP/112-1f3d 90443338182842@home: Up Dial(ZAP/34/3338182842|120|Tt) 2 active channels 1 active call And also the SIP/111 is always in Idle any idea of why ??? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051110/ca9568a2/attachment.htm
Hello How do you configure Polycom for presence please ? Harry --- Alvaro Parres <aparres@gmail.com> a ?crit :> Hi list, i have the next problem: > > I create 3 hints.. (111 (SIP/111), 112 (SIP/112), > and 102 (ZAP/35) ) > the SIP/111 is a GrandStream ATA > the SIP/112 is a Polycom 301 > the ZAP/35 is a Analogic Phone. > > The SIP/112 hints works great. But the other 2 no. > > The ZAP/35 is say is always in USE and as you see en > the > next console output is not in use. any Idea???? > > asterisk*CLI> > -= Registered Asterisk Dial Plan Hints =- > 111 : SIP/111 State:Idle Watchers 4 > 102 : ZAP/35 State:InUse Watchers 5 > 112 : SIP/112 State:InUse Watchers 2 > ---------------- > - 3 hints registered > asterisk*CLI> show cha > channel channels channeltypes > asterisk*CLI> show channels > Channel Location State Application(Data) > Zap/34-1 s@incel:1 Up Bridged Call(SIP/112-1f3d) > SIP/112-1f3d 90443338182842@home: Up > Dial(ZAP/34/3338182842|120|Tt) > 2 active channels > 1 active call > > And also the SIP/111 is always in Idle any idea of > why ??? > > thanks > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com > -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com