Asterisk integration with internet/outside world

Hi,

We have implemented Asterisk SIP server & SIP client Zoiper on different machines in our lab.
Lab network is a internal private network & its connected to switches which is connected to DSL box for internet. There is no firewall comes into the picture. DSL box is doing the natting.

How can the same SIP server is used by the SIP clients installed on machines with public IP outside internal LAN ? What are the configuration changes required for the below mentioned two cases ?
For example :-

Case 1 :-
A UAC on LAN wants to connect to UAS (Internet with Public IP) outside LAN.

Case 2 :-
A UAC on internet (Outside LAN) wants to connect to UAS on the internal LAN

Kindly suggest your valuable inputs.

Thanks !!
BR///Ankush Makkar

The primary documentation is in sip.conf.sample. You already know you are looking for NAT.

Note that it is almost certain that you have also implemented using Asterisk as a SIP client and the soft phone as SIP server, as you can’t have a useful two party phone call without that.

Thanks David.

I did the port forwarding in the DSL modem & now I am getting the register request.
But the problem is I am getting ACL error.

[Nov 11 19:18:44] NOTICE[4021]: chan_sip.c:26453 handle_request_register: Registration from ‘sip:demo-sgm@yy.yy.yy.yy;transport=UDP’ failed for ‘xx.xx.xx.xx:44256’ - Device does not match ACL

Do we need to define all client IP address in the sip.conf or this will work even without defining SIP client IP in the sip.conf ?

Thanks !!
BR///Ankush Makkar

Applying ACL is optional. If you don’t want to restrict addresses, don’t do so.

Thanks David.

Registration done from outside & SIP call is also setting up after disabling ACL.
Two Android SIP clients connected to the server & able to call each other but the problem is voice is one way. Directmedia is set to no (Remotely bridging) as both SIP clients are on internet.

Sample Logs :-

-- SIP/demo-anupam-00000011 is ringing
-- SIP/demo-anupam-00000011 answered SIP/demo-ashwani-00000010
-- [b]Remotely bridging SIP/demo-ashwani-00000010 and SIP/demo-anupam-00000011[/b]

[Nov 12 12:29:12] WARNING[4021]: chan_sip.c:4001 retrans_pkt: Retransmission timeout reached on transmission wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32001ms with no response
[Nov 12 12:29:12] WARNING[4021]: chan_sip.c:4030 retrans_pkt: Hanging up call wiki.asterisk.org/wiki/display/ … nsmissions).
== Spawn extension (users, 6002, 1) exited non-zero on ‘SIP/demo-ashwani-00000010’

=================================================================================
[Nov 12 12:11:12] WARNING[4021]: chan_sip.c:4001 retrans_pkt: Retransmission timeout reached on transmission 5bd92683458ffbba234a223f0e07f924@192.168.1.8:5060 for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 31999ms with no response
[Nov 12 12:11:35] WARNING[4021]: chan_sip.c:4001 retrans_pkt: Retransmission timeout reached on transmission wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 31999ms with no response
[Nov 12 12:11:35] WARNING[4021]: chan_sip.c:4030 retrans_pkt: Hanging up call wiki.asterisk.org/wiki/display/ … nsmissions).

Kindly suggest.

Thanks !!
BR///Ankush Makkar

Thanks David.

Tested with RTP locally bridging & My voice call is working.
Appreciate your support.

BR///Ankush Makkar