Port forward to two private IPs using port range

Created Jul 29, 2017 16:54:08Latest reply Oct 25, 2018 20:45:17 1575 4 0 0
  Rewarded E Coins: 0 (problem resolved)
Hi guys,

I'm trying to setup a SIP trunk connection throught an Huawei AR151 to a NEC SV9100 PBX system. The problem is that this NEC is using one IP address to register with the SIP server from the ISP and after that a second IP address to manage resources with that SIP server.

The ISP provided me with an IP address of 1.1.1.66 (made up) to connect with the SIP server and the NEC is using 172.16.0.10 for registering on port 5060 and 172.16.0.20 to manage resources on port range 10020 to 10400.

Another problem is that I'm getting the IP from the ISP not on the WAN side but on the LAN (and I have nothing phisically connected there) and I'm trying not to install a second router to perform NAT for the NEC system and use the Huawei AR151. To make the NAT work I've done this:

#I created an acl with the LAN from the NEC system to use for the NAT
acl number 2020
 description *** LAN PABX ***
 rule 10 permit source 172.16.0.0 0.0.255.255
#
# I created a pool from which NAT process would use an IP to NAT to
nat address-group 0 1.1.1..66 1.1.1.66
#
#
interface Ethernet0/0/4.899
 description *** Primary WAN link SIP ***
 dot1q termination vid 899
 ip binding vpn-instance VOICE
 ip address 2.2.2.45 255.255.255.0
 nat server protocol udp global 1.1.1.66 5060 inside 172.16.0.10 5060 vpn-instance VOICE
 nat outbound 2020
#

With this configuration I am able to dial in an IP phone behind the NEC system but the voice is not going thorught. I've tried to manually set some nat commands but it won't work.

I have managed to use a Cisco router after the Huawei and configured it to forward those port ranges and it works fine (of course a simple NAT router with portfowarding would do the job also but I didn't have one). I am still trying though to use just the AR151 but I can't manage to forward range of ports from the same inside global (1.1.1.2) to two differen inside local (172.16.0.10 and 172.16.0.20) IP addresses.

I have uploaded a network diagram to help better understand the situation. If you guys have any idea on how to work this out it would be greatly appreciated.

Thanks
BD




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No.9527  Mentor   Created Oct 25, 2018 20:45:17 Helpful(0) Helpful(0)

SIP AG Call Flow
1. The calling party POTS1 picks up a phone. When the SIP AG detects that the calling party picks up a phone, it allocates DSP resources to the calling party and plays a dial tone.
2. When detecting that the calling party dials the first digit of a dialing number, SIP AG1 stops the dial tone and matches the digit with a digitmap.
3. When detecting that the calling party dials the last digit of the dialing number, the SIP AG matches the dialing number with digitmaps. When the dialing number matches a digitmap, the SIP AG constructs an Invite message and sends it to the IMS proxy server.
4. The IMS proxy server receives the Invite message from the calling party, analyzes the called number, finds SIP AG2 to which the called party belongs, sends an Invite message to the called party, and returns a 100 Trying message to SIP AG1.
5. The called party receives the Invite message and responds with a 100 Trying message, indicating that it is receiving the called number.
6. SIP AG2 receives a complete called number, requests the called party POTS2 to ring, and sends a 180 Ringing message to SIPAG1. The calling party then hears the ringback tone.
7. The called party POTS2 picks up a phone. When SIP AG2 detects that the called party picks up a phone, it sends a 200 OK message to the calling party POTS1.
8. The calling party stops playing the ringback tone, and SIP AG1 sends an ACK message to the called party.
9. A call is established.
10. The calling party hangs up.
11. When detecting that the calling party hangs up the phone, the IMS proxy server sends a BYE message to the called party.
12. SIP AG2 receives the BYE message, plays the busy tone to the called party, and sends a 200 OK message to the calling party.
13. The called party hangs up the phone.

you can refer to below document to configure the SIP voice
http://support.huawei.com/hedex/pages/EDOC1000085855AEG05127/13/EDOC1000085855AEG05127/13/resources/dc/dc_cfg_voice_0061.html?ft=0&fe=10&hib=7.3.14.3.12.1&id=dc_cfg_voice_0061&text=Example%20for%20Configuring%20a%20SIP%20AG&docid=EDOC1000085855
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gululu  Admin   Created Jul 29, 2017 18:00:44 Helpful(0) Helpful(0)

@网络管理员Lemon please help
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Come on!
No.9527  Mentor   Created Oct 25, 2018 20:43:39 Helpful(0) Helpful(0)

SIP AG Call Flow
The calling party POTS1 picks up a phone. When the SIP AG detects that the calling party picks up a phone, it allocates DSP resources to the calling party and plays a dial tone.
When detecting that the calling party dials the first digit of a dialing number, SIP AG1 stops the dial tone and matches the digit with a digitmap.
When detecting that the calling party dials the last digit of the dialing number, the SIP AG matches the dialing number with digitmaps. When the dialing number matches a digitmap, the SIP AG constructs an Invite message and sends it to the IMS proxy server.
The IMS proxy server receives the Invite message from the calling party, analyzes the called number, finds SIP AG2 to which the called party belongs, sends an Invite message to the called party, and returns a 100 Trying message to SIP AG1.
The called party receives the Invite message and responds with a 100 Trying message, indicating that it is receiving the called number.
SIP AG2 receives a complete called number, requests the called party POTS2 to ring, and sends a 180 Ringing message to SIPAG1. The calling party then hears the ringback tone.
The called party POTS2 picks up a phone. When SIP AG2 detects that the called party picks up a phone, it sends a 200 OK message to the calling party POTS1.
The calling party stops playing the ringback tone, and SIP AG1 sends an ACK message to the called party.
A call is established.
The calling party hangs up.
When detecting that the calling party hangs up the phone, the IMS proxy server sends a BYE message to the called party.
SIP AG2 receives the BYE message, plays the busy tone to the called party, and sends a 200 OK message to the calling party.
The called party hangs up the phone.

you can refer to below document to configure the SIP voice
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No.9527  Mentor   Created Oct 25, 2018 20:43:50 Helpful(0) Helpful(0)

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