search for: anika_incoming

...iskdocs.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 at anika_incoming) exten => _03512222222,hint,SIP/00493511111111 exten => _03512222222,1,Verbose(2,Call for Anika...

...figure the pickup extension. (default is *8) ;pickupsound = beep ; to indicate a successful pickup (default: no sound) ;pickupfailsound = beeperr ; to indicate that the pickup failed (default: no sound) my users.conf: [general] callgroup = 1 pickupgroup = 1 my extensions.conf: [anika_incoming] exten => _00493512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _00493512222222,Set(CHANNEL(pickupgroup)=1) exten => _00493512222222,n,Dial(local/2222222 at anika_incoming) exten => _03512222222,1,Verbose(2,Call for Anika - [${CALLERID(num)}]) exten => _0351222222...

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > 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. Well, [anika_incoming] will be included in [default], of course... But I tried to define anika_incoming in subscribecontext, too. No changes... > Also, have you configured the phones as well? What do you mean? Thanks Luca Bertoncello (lucabert at lucabert.de)

Hi again! With the "call pickup"-function I can now pickup a call directed to another phone in my Asterisk. Very nice. My problem, now, is that I can't see on my phone, that the other phone (in another room) rings. Is it possible to signal the incoming call on other extension? I use two phones "Thomson ST2022". Thanks a lot Luca Bertoncello (lucabert at lucabert.de)

...4.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxanika] type=peer defaultuser=00493513333333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [messagenet] type=peer defaultuser=4444444444 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming...

...tmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxanika] type=peer defaultuser=00493513333333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [messagenet] type=peer defaultuser=4444444444 secret=MYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it po...

...=172.16.34.132 > port=5060 > fromuser=00493512222222 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [pbxanika] > type=peer > defaultuser=00493513333333 > secret= MYSECRET > dtmfmode=rfc2833 > host=172.16.34.132 > context=anika_incoming > outboundproxy=172.16.34.132 > port=5060 > fromuser=00493513333333 > fromdomain=172.16.34.132 > usereqphone=yes > canreinvite=no > insecure=invite > > [messagenet] > type=peer > defaultuser=4444444444 > secret=MYSECRET > dtmfmode=rfc2833 > host=sip.message...

Hi list! Right now I configured my Asterisk to forward the calls for the number X to both phones (mine and the phone of my wife). It works, of course, but I'm not enthusiast... I see what we have at office: if one phone rings, other phones in the same group can "catch the call", so that if a colleague is not present, another colleague can catch the call. I'd like to have the

...tmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493512222222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [pbxanika] type=peer defaultuser=00493513333333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493513333333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite [messagenet] type=peer defaultuser=4444444444 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.i...

...1) exited non-zero on 'SIP/00491773333333-0000000b' I tried to remove ALL includes in my [default], leaving just a log, but it calls, too... My [default] exten => _X.,1,Verbose(2,DEFAULT) include => internal_calls include => luca_incoming include => fax_incoming include => anika_incoming include => messagenet_incoming include => myproxy What's wrong, now? Many thanks for your help! Luca Bertoncello (lucabert at lucabert.de)

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone