Asterisk SIP trunk with 2 NIC

Asterisk Asterisk Support
1

Dear friends,

I have a problem with Asterisk Server in connect to SIP Provider. Please help me. Thanks.
Here are information about Asterisk Server below :

  • Asterisk has 2 NIC : eth0 with IP 10.0.0.2 , eth1 with IP 10.165.1.30

  • eth0 with NAT (internet FTTH), used for many SIP client with NAT mode.

  • eth1 with no NAT (used leased line) to connect SIP Provider with SIP Trunking (peer-to-peer).

  • For local calls : internal calls is OK (configure in SIP setting with NAT = yes, localnet = 10.0.0.0/255.255.0.0, externip = 118.69.68.15)

  • For PSTN : call in without voice, call out hangup with error “488 = Not Acceptable Here” from Provider Server.

  • About routing IP (layer 3) : routing with 2 NIC for Asterisk is OK (check with ping, tracert, and SIP message debug).

  • About SIP connection (layer 7): After check debug, i found that a problem in Contact in SIP message from Asterisk.

  • Contact: sip:19001234[color=#BF4000]@118.69.68.15 [/color]in SDP session. So the SIP Provider deny calls because of the wrong IP in Contact. Now, i want to change SIP Message Contact sip:19001234@10.165.1.30 to the provider accept calls. However, i don’t know how to change the information. Please help me soon.

PS : when i have tried to remove NAT in SIP setting , call in/call out with the provider is OK (because this time Contact in SIP Signalling will be sip:19001234@10.165.1.30). But with this configuration, SIP clients in WAN is fail. So, asterisk server can’t remove NAT in eth0.

2

Please provide your sip.conf. nat= probably doesn’t do what you think it does.

3

hi devid55,

Thank you for your reply. Here is the information i configured in sip.conf :

nat = yes
localnet= 10.0.0.0/255.255.0.0
fromdomain=voip-xx.dyndns.org

I used DynDNS service to map to IP 118.69.68.15.

4

nat=yes is deprecated, use nat=force_rport,comedia instead

5

Dear ambiorixg12,

My asterisk server is installed with asterisk 1.6, freePBX 2.9. So, there is NO options “nat=force_rport” in FreePBX to choose.

NAT has 4 options : Yes, No, Never, Route
Please see this Picture :

6

You are using a very old version of both softwares!

7

Use nat= instead. On reason for deprecation is probably that requesting everything is not what is wanted most times.

8

As well as using unsupported software, you haven’t told Asterisk what your public address is. Asterisk is too old to support. FeePBX isn’t supported here.

9

Hi David55,

I will upgrade my Asterisk System. Please recommend for me about a suitable version of Asterisk and FreePBX to fix the problem above.

Thanks & Best Regards,

10

The problem is almost certainly a configuration error, although I would note that improper dual homed systems are never going to be easy (i.e. ones without autonomous system numbers). For new installs of Asterisk you should use the latest non-beta/non-release candidate minor version of Asterisk 13. We have no knowledge of FreePBX.

11

I’m expecting to see externip, externhost, or stunaddr in the configuration.

12

hi David55,

I add : externip=118.69.68.15
I don’t use stun server. I also coundn’t use stun because have to record all of calls.

Best Regards,

13

hi friends,

The problem was SOLVED. That is simple.
I have 2 steps to setup :

  • Route SIP Provider IP range to go in eth1.
  • Add localnet for SIP Provider (add localnet for eth0, eth1 is NOT enough).