Hi No idea ?
thanks Olivier Le 24 avril 2012 16:06, Olivier CALVANO <[email protected]> a écrit : > Hi > > i have a strange problems on my asterisk server: > > I have two asterisk server. > > On the first, i use realtime with a MySQL Database, > i have two user: > USER01 > USER02 > exactly the same configuration only username and password has different. > > > On my second server (phone is connected on this server): > > I have in sip.conf: > > register => USER01:[email protected]/USER01 > register => USER02:[email protected]/USER02 > > [USER01] > type=friend > username=USER01 > secret=1234 > host=172.16.0.11 > qualify=yes > dtmf=rfc2833 > nat=no > canreinvite=no > canredirect=no > dtmfmode=rfc2833 > disallow=all > allow=alaw > context=I-User01 > musiconhold=default > insecure=port,invite > > [USER02] > type=friend > username=USER02 > secret=5678 > host=172.16.0.11 > qualify=yes > dtmf=rfc2833 > nat=no > canreinvite=no > canredirect=no > dtmfmode=rfc2833 > disallow=all > allow=alaw > context=I-User01 > musiconhold=default > insecure=port,invite > > > i see the registration: > > ipbx*CLI> sip show registry > Host dnsmgr Username Refresh State > Reg.Time > 172.16.0.11:5060 N USER01 105 Registered > Tue, 24 Apr 2012 15:58:58 > 172.16.0.11:5060 N USER02 105 Registered > Tue, 24 Apr 2012 15:58:59 > > > > > i have one phone connected to the context "I-User01" and another > connected to "I-User02" > > When i call with the phone connected to I-User01, no problems, that's > work but when i call > with the second phone (use I-User02) i have a error: > > > On the first server: > [Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username > mismatch, have <USER01>, digest has <USER02> > [Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096 > handle_request_invite: Failed to authenticate device "Olivier" > <sip:[email protected]>;tag=as0cd775ab > > > The exten: > > On I-User01: exten => _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r) > On I-User02: exten => _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r) > > > > i i change on the I-User02: > Dial(SIP/USER02/${EXTEN:1},90,r) > in > Dial(SIP/USER01/${EXTEN:1},90,r) > all call work's. > > > anyone have a idea ? i think's that i have a error but don't see where > > best regards > Olivier -- _____________________________________________________________________ -- 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
