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
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Sorry, forgot to mention what ports the Asterisk RTP protocol contains
name: RTP
Parameters:
Primay connections:
- 8000-10000 - UDP - Recieve Send
Secondary connections:
- 8000-10000 - UDP - Send Receive
And here is my RTP.CONF
rtpstart=8000
rtpend=10000
And here my SIP.CONF
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
externip=<external ISA>
nat=yes
localnet=192.168.30.0/255.
Like you can see in my previous posts, these ports are already forwarded:
RTP
Primary connections:
- 8000-10000 - UDP - Recieve Send
Secondary connections:
- 8000-10000 - UDP - Send Receive
SIP
Primary connections:
- 5060-5082 - TCP - Inbound
- 5060-5082 - UDP - Recieve Send
Secondary connections:
- 5060-5082 - TCP - Outbound
- 5060-5082 - UDP - Send Receive
But on the ISA Server the error 0x80070034 ERROR_DUP_NAME occurs with the RTP rule in the logs.
1. I use Linux for firewalling, and haven't used ISA in quite a while, but if you can, try eliminating all the secondary connections.
HRESULT_FROM_WIN32(ERROR_D
The call failed because the filter previously called AllowFutureConnect to allow a connection to the same IP address and port.
http://msdn2.microsoft.com
2. Asterisk doesn't like being behind a NAT / Firewall. The easiest way to accomplish what you need is to put the box in a DMZ.
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Answer for Membership
by: grbladesPosted on 2008-03-15 at 10:19:33ID: 21133467
When using Asterisk you will need to make sure the following ports are redirected to the asterisk server.
SIP - UDP 5060
RTP - UDP 10000-20000
Then within asterisk itself you need to define the l'localip' and either the 'externip' or 'externhost' within sip.conf. This is so that when asterisk detects a call coming from the internet it uses the external IP address within the SIP packets.