Hi list: I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint extension. But they are not working: /etc/asterisk/extensions.conf .... [sip-test] exten => 116,1,dial(SIP/116) exten => 112,hint,SIP/112 exten => 112,1,dial(SIP/112) .... /etc/asterisk/sip.conf ... [112] type=friend ; either "friend" (peer+user), "peer" or "user" context=sip-test host=dynamic ; we have a static but private IP address nat=no ; there is not NAT between phone and Asterisk canreinvite=yes ; allow RTP voice traffic to bypass Asterisk dtmfmode=RFC2833 ; either RFC2833 or INFO for the BudgeTone incominglimit=2 ; permit only 1 outgoing call at a time callgroup=2 ; callgroup pickupgroup=2 ; pickupgroup mailbox=112@default ; mailbox 1234 in voicemail context "default" allow=all ; need to disallow=all before we can use allowlanguage=es progressinband=no qualify=yes .... when at the console i send a show hints i get this: -= Registered Asterisk Dial Plan Hints =- 112 : SIP/112 State:Unknown Watchers 1 ---------------- - 1 hints registered any idea ??? Thanks. ALvaro Parres. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051109/b67db630/attachment.htm
Pls update to rc1 and if you're still having the same problem, open a bug on bugs.digium.com so we can get it fixed before release. On 11/9/05, Alvaro Parres <aparres@gmail.com> wrote:> Hi list: > > I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint > extension. But they are not working: > > > /etc/asterisk/extensions.conf > .... > [sip-test] > exten => 116,1,dial(SIP/116) > exten => 112,hint,SIP/112 > exten => 112,1,dial(SIP/112) > .... > > /etc/asterisk/sip.conf > ... > [112] > type=friend ; either "friend" (peer+user), > "peer" or "user" > context=sip-test > host=dynamic ; we have a static but private IP > address > nat=no ; there is not NAT between phone > and Asterisk > canreinvite=yes ; allow RTP voice traffic to bypass > Asterisk > dtmfmode=RFC2833 ; either > RFC2833 or INFO for the BudgeTone > incominglimit=2 ; permit only 1 outgoing call at a > time > callgroup=2 ; callgroup > pickupgroup=2 ; pickupgroup > mailbox=112@default ; mailbox 1234 in voicemail context > "default" > allow=all ; need to disallow=all before we > can use allow> language=es > progressinband=no > qualify=yes > .... > > when at the console i send a show hints i get this: > > -= Registered Asterisk Dial Plan Hints =- > 112 : SIP/112 State:Unknown > Watchers 1 > ---------------- > - 1 hints registered > > > any idea ??? > > Thanks. > > ALvaro Parres. > > > > _______________________________________________ > --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 > >
Upgrade to asterisk-1.2.0-rc1 and ensure that your sip file contains subscribecontext=sip-text.
Thanks with the upgrade they work... Now i only have one 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 On 11/9/05, Peter Dean <peter.john.dean@gmail.com> wrote:> > Upgrade to asterisk-1.2.0-rc1 and ensure that your sip file contains > subscribecontext=sip-text. > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com <http://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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051110/6552c24c/attachment.htm