Im running an AsteriskNOW server on my internal network (192.168.30.12) and an ISA 2006 server that provides the internet for the internal clients and servers (ISA internal: 192.168.30.10).
Now I want to make it possible for external users (VoIP phones) to connect to my internal Asterisk server using SIP.
Situation:
AsteriskNOW SIP server ---- NAT ISA server > Internet < Cisco NAT router ---- Linksys Voip Phone
I already published the following firewall rules:
1)
name: Asterisk SIP
Action: Allow
Traffic: Protocol: SIP
From: Anywhere
To: 192.168.30.12 ( * request appears to come from original client)
Networks: All networks
2)
name: Asterisk RTP
Action: Allow
Traffic: Protocol: RTP
From: Anywhere
To: 192.168.30.12 ( * request appears to come from original client)
Networks: External
Protocol information:
name: SIP
Parameters:
Primary connections:
- 5060-5082 - TCP - Inbound
- 5060-5082 - UDP - Recieve Send
Secondary connections:
- 5060-5082 - TCP - Outbound
- 5060-5082 - UDP - Send Receive
The IP Phone can register itself with Asterisk but when I trie calling the voicemail number (or another SIP phone) on the asterisk there is no sound.
When I log traffic with ISA, this appears several times when dialing the voicemail number:
Original Client IP <Internet IP of ip phone>
Client IP <Internet IP of ip phone>
Destination IP 192.168.30.12
Protocol RTP
Transport UDP
Source Network External
Destination Network Internal
Action Failed Connection Attempt
Rule Asterisk RTP
Log Time 15/03/2008 13:36
Source Port 16388
Destination Port 9824
Processing Time 0
Bytes Sent 0
Bytes Received 0
Result Code 0x80070034 ERROR_DUP_NAME
In the Cisco router of the external client that tries to connect, the following Nat Translation is visible when dialing the voice mail number.
Protocol udp
Inside global <Internet IP of ip phone>:16399
Inside local 190.168.1.40:16388
Outside local <external ISA>:9824
Outside global <external ISA>:9824
So what does the ERROR_DUP_NAME error means and how can I solve my problem?
Thanks in advance
