I have a GS101 connected to * with sip and g729. When an incoming call comes in from outside (via pstn for example), and no one picks up the GS, * reports that 'the user is on the phone'. If no one answers, I'd expect it to report 'unavailable'. Maybe I'm not understanding the call flow ... (should it be u$ at '2', then b$ at '102' ?) My current config for call flow seems to match others I've seen on the wiki, etc. my extensions.conf for the grandstream at x2015 - [incoming] exten => 2015,1,Dial(SIP/2015@2015,20,T,t) exten => 2015,2,Voicemail(b${EXTEN}) exten => 2015,3,Hangup exten => 2015,102,Voicemail(u${EXTEN}) exten => 2015,103,Hangup Thanks, Chris Clifton
The right conf must be like this: exten => 2015,1,Dial(SIP/2015@2015,20,T,t) exten => 2015,2,Voicemail(u${EXTEN}) exten => 2015,102,Voicemail(b${EXTEN}) exten => 2015,103,Hangupv Chris HARIGA Techselesta Inc. http://www.techselesta.com/ ----- Original Message ----- From: "Chris Clifton" <chris@netlabz.com> To: <asterisk-users@lists.digium.com> Sent: Tuesday, March 02, 2004 10:28 PM Subject: [Asterisk-Users] gs on phone ?> > I have a GS101 connected to * with sip and g729. > > When an incoming call comes in from outside (via pstn for example), and no > one picks up the GS, * reports that 'the user is on the phone'. If no one > answers, I'd expect it to report 'unavailable'. > > Maybe I'm not understanding the call flow ... (should it be u$ at '2',then> b$ at '102' ?) My current config for call flow seems to match others I've > seen on the wiki, etc. > > my extensions.conf for the grandstream at x2015 - > > [incoming] > exten => 2015,1,Dial(SIP/2015@2015,20,T,t) > exten => 2015,2,Voicemail(b${EXTEN}) > exten => 2015,3,Hangup > exten => 2015,102,Voicemail(u${EXTEN}) > exten => 2015,103,Hangup > > Thanks, > Chris Clifton > > _______________________________________________ > 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 >
In your extensions.conf, the "b" and "u" are reversed. Use u${EXTEN} for priority 2 and b${EXTEN} for priority 102. -Ron -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Chris Clifton Sent: Tuesday, March 02, 2004 10:29 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] gs on phone ? I have a GS101 connected to * with sip and g729. When an incoming call comes in from outside (via pstn for example), and no one picks up the GS, * reports that 'the user is on the phone'. If no one answers, I'd expect it to report 'unavailable'. Maybe I'm not understanding the call flow ... (should it be u$ at '2', then b$ at '102' ?) My current config for call flow seems to match others I've seen on the wiki, etc. my extensions.conf for the grandstream at x2015 - [incoming] exten => 2015,1,Dial(SIP/2015@2015,20,T,t) exten => 2015,2,Voicemail(b${EXTEN}) exten => 2015,3,Hangup exten => 2015,102,Voicemail(u${EXTEN}) exten => 2015,103,Hangup Thanks, Chris Clifton _______________________________________________ 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