How to divert a VOIP trunk ?

henkoegema
Joined: Sun 25 of Dec, 2005

Re: How to divert a VOIP trunk ?

Posted:Tue 09 of Sep, 2008 (18:24 UTC)
In Re: How to divert a VOIP trunk ? (Sun 07 of Sep, 2008 (22:46 UTC))mjfischerwrote:
You should probably try to use a port sniffer like “Ethereal” and then place an inbound call. Monitor the ROUTER IP address to see if the packets are actually going to the “192.168.1.101” address. With this limited information it really sounds like a router issue so far. If you gather more information we could probably get you though this pretty quickly.

Also can you currently accept an inbound call to the FreeSWITCH on the “192.168.1.101” address from an IP phone..??? Make sure of this before you re-route your traffic..!


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I've installed Wireshark on a pc with ip address 192.168.1.101 (using Ubuntu 8.04)

How do I capture all traffic from my router (192.168.1.1) ?

I've used as filter ip.adress == 192.168.1.1 but that doesn't work.

Henk
henkoegema
Joined: Sun 25 of Dec, 2005

Re: How to divert a VOIP trunk ?

Posted:Tue 09 of Sep, 2008 (13:47 UTC)
I will try your suggestion.

I have several phones connected to FS and they can all call eachother.
Can also call from Asterisk to FS and v.v

Only VOIP trunk calls and can't route to FS.

Henk
mjfischer
Joined: Sun 07 of Sep, 2008

Re: How to divert a VOIP trunk ?

Posted:Sun 07 of Sep, 2008 (22:46 UTC)
You should probably try to use a port sniffer like “Ethereal” and then place an inbound call. Monitor the ROUTER IP address to see if the packets are actually going to the “192.168.1.101” address. With this limited information it really sounds like a router issue so far. If you gather more information we could probably get you though this pretty quickly.

Also can you currently accept an inbound call to the FreeSWITCH on the “192.168.1.101” address from an IP phone..??? Make sure of this before you re-route your traffic..!
henkoegema
Joined: Sun 25 of Dec, 2005

How to divert a VOIP trunk ?

Posted:Sun 07 of Sep, 2008 (13:37 UTC)
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