2012/4/25 Olivier CALVANO <o.calv...@gmail.com> > Sure, sorry for the Confusion ;=) > > > > > Server A "Trader": > Asterisk Server 1.6.x for call routing only. > IP Adress: 172.16.0.11 > Use Realtim on MySQL Database > This server route all call to a lot of VoIP Carrier. > > > Server B "Ipbx" > Asterisk Server 1.6.x for connect a lot of Linksys SPA942 phone. > IP Adress: 172.16.0.70 > Second IP: 172.16.1.70 (used for phone lan) > Use Realtim on MySQL Database > This server route all call to a lot of VoIP Carrier. > > > Linksys SPA942 A: > IP Adress: 172.16.1.200 > Connected in SIP at Server B IPBX > use sip.conf (no realtime) > context: I-User01 > > > Linksys SPA942 B: > IP Adress: 172.16.1.220 > Connected in SIP at Server B IPBX > use sip.conf (no realtime) > context: I-User02 > > > > On Server A "Trader", we have two sip account: > accountname: "USER01" for user in group 1 > accountname: "USER02" for user in group 2 > > On Server B "Ipbx", i use registry: > register => USER01:1234@172.16.0.11/USER01 > register => USER02:5678@172.16.0.11/USER02 > for two connection to the Trader Server. Registry is good: > on server A "Trader": > > trader*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 > > > On server B "Ipbx", i have into my sip.conf after the registry: > > [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 > > and in extensions.conf: > > [I-User01] > exten => _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r) > > [I-User02] > exten => _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r) > > > > > > > > When i call with Linksys SPA942 A, i use the context "I-User01" and > the call are sent > to SIP account "USER01" and No problems. > > When i call with Linksys SPA942 B, i use the context "I-User02" and > the call are sent > to SIP account "USER02" but Server A "Trader" reject the call > immediatly with this error: > > [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:906280@172.16.0.70>;tag=as0cd775ab > > "Olivier" and "906280" is the information that i have on the Linksys > SPA942 B, 906280 is the username used between > > > > > best ? hihi > Olivier > > > > > > Le 25 avril 2012 06:38, SamyGo <govoi...@gmail.com> a écrit : > > Hi, > > Lots of mixing and confusing stuff - Can you re-explain the topology you > are > > trying to achieve with proper IP addresses and declared sip ext. names. > > > >> 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: > > > > > > Somehow it reminds of the same situation I always face when a peer is > > declared with the same name as of the dialing one on second server - only > > Its just not registered there instead registered on server-1. > > So when the call comes in from server-1 to server-2 fromuser=olivier > which > > is not registered on server-2 but is declared. Server-2 thinks that this > is > > my valid extension but it is not registered with me and so lets > authenticate > > this one and here it fails and rejects the call. > > > > BR, > > Sammy. > > > > On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO <o.calv...@gmail.com> > > wrote: > >> > >> 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 => USER01:1234@172.16.0.11/USER01 > >> register => USER02:5678@172.16.0.11/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:906280@172.16.0.70>;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 > >> > >> -- > >> __ >
Remove the "insecure=invite,port" and maybe add the match_auth_username=yes in the sip.conf general section Leandro
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