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?
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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
unplug wrote:
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.
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
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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@pernau.at wrote:
unplug wrote:
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.
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
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@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@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
-
Klaus Darilion -
unplug