On 30 December 2015 at 10:03, Luca Bertoncello <[email protected]> wrote:
> Ishfaq Malik <[email protected]> schrieb: > > Hi Ishfaq > > > Look into Busy Lamp Field/Presence > > > > Here's a starting point: > > > > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html > > Thanks a lot, but it does not seems to work... > > Here my configuration: > > sip.conf: > > [general] > allowsubscribe=yes > subscribecontext = default > notifyringing = yes > notifycid = yes > callcounter = yes > > extensions.conf: > > [anika_incoming] > exten => _00493512222222,hint,SIP/00493511111111 > exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) > exten => _00493512222222,n,Dial(local/2222222@anika_incoming) > exten => _03512222222,hint,SIP/00493511111111 > exten => _03512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) > exten => _03512222222,n,Dial(local/2222222@anika_incoming) > exten => _2222222,hint,SIP/00493511111111 > exten => _2222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) > exten => _2222222,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = > "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})}) ; Damit das "+49" mit "0" > ersetzt wird > exten => _2222222,n,Set(CHANNEL(musicclass)=default) > exten => _2222222,n,Dial(SIP/00493512222222,19,RcxX) > exten => _2222222,n,Verbose(2,Voicemail for Anika) > exten => _2222222,n,Set(CALLERID(name)=) > ; Damit in der E-Mail der AB nicht den Namen steht > exten => _2222222,n,VoiceMail(00493512222222,us) > exten => _2222222,n,Hangup > > then I reloaded the core (core reload), SIP (sip reload) and Dialplan > (dialplan reload) and I called the 03512222222 from my mobile phone. > It rings, but on the other phone (03511111111) is nothing to see... > > Where is my error? > > Thanks > Luca Bertoncello > ([email protected]) > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > The hints have to be in the same contexts in extensions.conf as defines in the sip.conf subscribecontext which can be set per peer. Also, have you configured the phones as well? -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)161 660 2350 f: +44 (0)161 660 9825 e: [email protected] w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
