11 Dec
2005
11 Dec
'05
3:39 p.m.
Below is the common configuration of the network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
12 Dec
12 Dec
11:52 a.m.
Use ngrep to watch for incoming SIP requests on the SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
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Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
13 Dec
13 Dec
4:49 a.m.
From the log, it showed lookup(): 'account
name' Not found in usrloc.
I think it is the NAT problem, so I use stun for the
case below.
Telephone (A)
|
PSTN
|
G/W
|
openser
|
NAT -- IP phone(B) -- STUN
+ ----- IP phone(C) ------+
Below is the result:
A to B/C is ok
B/C to A is ok
However, there is no sound when B to C or vice versa. What reason
will cause no sound between B and C? Is the the reason from the
NAT/STUN?
On 12/12/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
Use ngrep to watch for incoming SIP requests on the
SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the
network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
12:43 p.m.
unplug wrote:
Does the INVITE from the GW does have a proper formated request URI? The
called phone number must be mapped to the username before doing
lookup("location"). E.g. you can use aliases to map phone numbers to
user names lookup("aliases"); (or use the aliasdb module)
regards
klaus
From the log,
it showed lookup(): 'account name' Not found in usrloc.
I think it is the
NAT problem, so I use stun for the case below.
Telephone (A)
|
PSTN
|
G/W
|
openser
|
NAT -- IP phone(B) -- STUN
+ ----- IP phone(C) ------+
Below is the result:
A to B/C is ok
B/C to A is ok
However, there is no sound when B to C or vice versa. What reason
will cause no sound between B and C? Is the the reason from the
NAT/STUN?
On 12/12/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
Use ngrep to watch for incoming SIP requests on
the SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
14 Dec
14 Dec
9:45 a.m.
I think that it is a NAT problem. Found from some document showing
that using STUN is not stable much. As I know, openser will have a
nathelper module to process the NAT. There is an configuration
example in the source "nathelper.cfg". Does it work to solve the NAT
problem instead of using STUN?
On 12/13/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
unplug wrote:
Does the INVITE from the GW does have a proper formated request URI? The
called phone number must be mapped to the username before doing
lookup("location"). E.g. you can use aliases to map phone numbers to
user names lookup("aliases"); (or use the aliasdb module)
regards
klaus
From the
log, it showed lookup(): 'account name' Not found in usrloc.
I think it is
the NAT problem, so I use stun for the case below.
Telephone (A)
|
PSTN
|
G/W
|
openser
|
NAT -- IP phone(B) -- STUN
+ ----- IP phone(C) ------+
Below is the result:
A to B/C is ok
B/C to A is ok
However, there is no sound when B to C or vice versa. What reason
will cause no sound between B and C? Is the the reason from the
NAT/STUN?
On 12/12/05, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote:
Use ngrep to watch for incoming SIP requests on
the SIP proxy.
Take a look at the logfiles on the gateway.
klaus
unplug wrote:
Below is the common configuration of the network.
Telephone -- PSTN -- G/W -- openser -- softphone (eg windows messenger)
I can make call from softphone to Telephone. However, it is failed to
make call from Telephone to softphone. I wonder why it happened and
any reference to trace the problem. Anyone have such experience?
_______________________________________________
Users mailing list
Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
6922
days inactive
6925
days old
4 comments
2 participants
participants (2)
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Klaus Darilion -
unplug