How to divert a virtual PSTN line to another server ?
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
I'm still looking for a solution. :working: henkoegema wrote:
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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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:
I'm still looking for a solution. :working:
henkoegema wrote:
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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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:
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?
-- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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:
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:
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?
-- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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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:
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:
-- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---... Sent from the Freeswitch-users mailing list archive at Nabble.com.
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 -- View this message in context: http://www.nabble.com/How-to-divert-a-virtual-PSTN-line-to-another-server---... Sent from the Freeswitch-users mailing list archive at Nabble.com.
participants (4)
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Henk Oegema
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henkoegema
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Ivan C Myrvold
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Robert Dyck