[asterisk-users] Strange problem on ougoing call
Leandro Dardini
ldardini at gmail.com
Wed Apr 25 02:34:27 CDT 2012
2012/4/25 Olivier CALVANO <o.calvano at 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 at 172.16.0.11/USER01
> register => USER02:5678 at 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 at 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 <govoiper at 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.calvano at 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 at 172.16.0.11/USER01
> >> register => USER02:5678 at 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 at 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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