[asterisk-users] Strange problem on ougoing call

Wed Apr 25 02:20:02 CDT 2012

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
>>
>> --
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>
>
>
> --
> _____________________________________________________________________
> -- 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