Topic: One Last Gremlin! EXT -> REMOTE EXT and Vice Versa

[the_owl]

Hi All!!

I have, with your help, almost completed my first * installation!! - just one last gremlin to beat...

This is the setup

> One office with the sme/astersk/sail server and five (Local) extentions

> Five brances each with one (remote) extention

> Only for EXT to EXT calls, no trunks involved, but maybe later?

All branches have been equipped with Thomson Speedtouch ST716 routers with the extention phone on the VOIP ports - they have been configured as remote and are registering with the server - this is confirmed by the server and the router config interface - they can dial in and hear system messages (time, ext No.,) but the echo test always cuts out at "to end the test...Silence" and can make other phones ring but get silence in both directions . however if the phone is left to ring we hear "the person at EXT No. X is unavailable... etc"

At the office I have configured the same routers as ATA boxes using a TPL provided by the supplier.  These also register perfectly and can make calls within the office but when they call the branches the line rings but the get silence in both directions upon answer.

The server setup at the office is

ADSL -> Router (firewall disabled, DDNS on and working, DMZ to server static IP 192.168.0.3) -> Server (192.168.10.1) -> LAN

After reading through this forum I think (remember I'm new to this so PLEASE correct me if I'm wrong!) it is one of four things:

> The external IP in globals, currently set to 192.168.0.3.
I have tried putting the DDNS address in there as well but it cuts the phones off

> SME Firewall blocking SIP traffic - Not sure how to check

> Something in the SIP Header file, currently "localnet" is set to 102.168.0.3, also tried putting 192.168.10.1 in there, nothing seemed to happen?

> the right answer from you fine people!!

The Organisation may be moving all the broadband connections to a new provider with static IP's in the near future, but this is not confirmed at the moment.

Can anyone please tell me where I've gone wrong?

Many Thanks In advance

[SARK devs]

local net is a mask it should be set to nnn.nnn.nnn.0
externalip should be your true external IP (if you don't know what it is then you can get it from "what is my ip addess" in google)
Your remore extensions (i.e. outside the local subnet) should be set to "remote" in the extension definition.

If all of these things are OK and you still can't flow voice then it is probabaly a router/firewall problem.  Your near-end router should flow 5060 and 10000-20000 (all UDP) to your SAIL box.  If your SAIL box is server only then you can shut down the local firewall with /etc/init.d/masq stop.  DO NOT ATTEMPT TO SHUTDOWN THE FIREWALL IF YOU ARE RUNNING SERVER_GATEWAY!!

If it still isn'y flowing voice then post back here.

Kind Regards

S

[the_owl]

Thanks for the feedback S!

And for tha amazing speed of the response!

The server is in server-gateway mode, The router is set to blindly send all traffic to the server's WAN interface via DMZ, with any firewall in the router disabled.


Can I just check I'm correct in this?

Localnet=

192.168.0.0 (pointing to the router then net)

Thanks Again!

Many thanks

JUST THINKING ABOUT THIS!!

I'm currently using DDNS, all the offices are connecting using XXXXXXXX.dyndns.org to get to the server - could this cause a problem??

[the_owl]

Initial tests indicate that this has SOLVED the problem!!

I am now able to call between remote extentions, I will be testing calls Local -> Remote tomorrow. It's now 10:30pm here!

For the benefit of other souls looking for answers heres how I implimented the advice given:

I changed the localnet setting in Headers > sip.conf > to 192.168.10.0/255.255.255.0
I then changed the external ip in Global Settings to xxxxxx.dyndns.org

It now seems to be working - fingers crossed for the morning!

I will post with any developments

[David Harper]

Although it's just personal preference, I always recommend that business users have a static IP address and their own domain name. This keeps things nice and simple, and usually doesn't cost a lot extra.

[the_owl]

Hi David,

I fully agree, I have already discussed this with the client and they are planning to move forward once they see the system up and running.

The asterisk side is just one part of the project, however I'm dealing with people who , although open to the idea, seem to be reluctant to believe that it will do what has been suggested - in short the are waiting to taste the pudding!

After some delays on other jobs today i didn't get to the office to setup and test the phones there - this is in the diary now for tomorrow and I will report back for the benefit of others who may be interested.

Until then!

[the_owl]

UPDATE AS PROMISED!!!

I Got to the clients head office over the weekend to setup the phones there, plugged in and worked first time!!

Branches are now in full communication through the Asterix system! - not sure if the owner is pleased or worried with this new found capability!

A Big thanks to the developers, of this system, and the many experts who take the time to share their knowledge with the rest of us!