Re: [asterisk-users] Strange problem on ougoing call
Perfect that's work ;=)
very thanks
Le 25 avril 2012 10:19, Olivier CALVANO o.calv...@gmail.com a écrit :
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
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
Re: [asterisk-users] Strange problem on ougoing call
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
Re: [asterisk-users] Strange problem on ougoing call
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
Re: [asterisk-users] Strange problem on ougoing call
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
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
Re: [asterisk-users] Strange problem on ougoing call
Hi
No idea ?
thanks
Olivier
Le 24 avril 2012 16:06, Olivier CALVANO o.calv...@gmail.com 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 = USER01:1234@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
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Re: [asterisk-users] Strange problem on ougoing call
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.comwrote:
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
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