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[pabx_freeswitch at tel...]

I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as
incoming external line to my Asterisk server (192.168.1.100)

In my router I have have :
Application Start End Protocol IP Address
-----------------------------------------------------------------------------
SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.100
RTP 5090 to 5100 UDP 192.168.1.100


That works OK.


Now I want to divert that PSTN line from Asterisk to my Freeswitch server
(192.168.1.101)
So I changed in my router the ip addreese from 192.168.1.100 to 192.168.1.101

Application Start End Protocol IP Address
-----------------------------------------------------------------------------
SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.101
RTP 5090 to 5100 UDP 192.168.1.101


But.....when an external call comes in, it still goes to Asterisk.

Am I on the wrong track or ....... (?)

Rgds
Henk




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[pabx_freeswitch at tel...]

I'm still looking for a solution. :working:



henkoegema wrote:

[rob.dyck at telus.net]

If your port forwarding is not working perhaps you should seek technical
support from the manufacturer of your particular router. Is there a forum for
this router?

On Sunday 07 September 2008, henkoegema wrote:

[pabx_freeswitch at tel...]

I have not experienced any problem with port-forwarding.

Is my problem a port-forwarding problem or could it be something else.
:confused:



Robert Dyck wrote:

[ivan at myrvold.org]

If your call ends up at the IP address where Asterisk is running,
clearly your port forwarding is not working. Try to flip the ip
addresses of the Asterisk and FreeSWITCH, and see if it now ends up at
the FreeSWITCH.

Ivan

Den 8. sep.. 2008 kl. 12:03 skrev henkoegema:

[pabx_freeswitch at tel...]

It's getting more mysterious to me. :confused:

If have flipped Asterisk and FS.
Asterisk is now at 192.168.1.101 and
FS is now at 192.168.1.100

Port-forwarding is: SIP 5004-5082 UDP/TCP 192.168.1.100

But incoming calls STILL go to Asterisk. :confused: Which it
shoudn't.

When I shutdown Asterisk, I get not-reachable (?) tone. (from my mobile)

Henk



Ivan C Myrvold wrote:

[pabx_freeswitch at tel...]

To see what's happening with the voip packets from my router I want to do
following:

I have Wireshark installed on pc 192.168.1.101.

Can (if so, how?) I monitor traffic from my router (192.168.1.1) and
flowing into the LAN network (f.e. to 192.168.1.100) ?

Or can I only monitor traffic from and to 192.168.101 (where Wireshark is
installed) ?


Henk
--
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