[Harinda]

I have my asterisk box in a separate location to the SPA 3K,Asterisk box has static IP and SPA 3K with a dynamic IP.

I want to send PSTN calls out using the SPA 3K , any suggestions will be greatly appreciated .

[hwittenb]

For the SPA3102, I would get a symbolic address from someone like DynDns and devise a means to keep the address up to date for the ip address translation (router support for dynamic dns or program on a pc behind the router).

I would then setup the voip-to-pstn gateway for http digest authentication, setup a userid and password for the http authentication. Set answer call without registration on the PSTN Line.

With the Asterisk system setup the trunk with the symbolic address and the port number for the pstn line. Setup the user id and password that you created above. Setup the asterisk trunk not to register. An asterisk system user will dial the pstn number. You route the dialed pstn number to this trunk. The Asterisk system will send a sip invite to the SPA3102 and the SPA3102 will dial the outgoing number.

Edit: An alternative would be to setup the SPA3102 PSTN Line as an extension on the Asterisk system and someone would call the extension and then get a dial tone to dial out. If you did it this way you wouldn't have to deal with the dynamic address because the registration process would take care of that problem.

[Harinda]

Thanks for your reply , actually I'm using SIPURA SPA 3000 , sorry for not mentioning earlier. Anyway I got your idea and will update the result.

[markosjal]

I had a similar issue with the SPA3102 that was on a DSL modem and Dynamic IP, and Trixbox on a separate Fixed IP. Since there was no computer there nor did the SPA have a DYNDNS client (which I see as an oversight in the SPA3102), I devised the following solution:

1) Register the PSTN line to asterisk as you would any other device. For this example we will say it is registered as extension 300

2) Create a custom trunk with custom dial string: SIP/$OUTNUM$@300

3) Create outbound route pointing to the custom trunk

I looked high and low for a work around to a fixed IP or a DYNDNS , and never found much. This solution has now worked flawlessly for 6+ months.

Mark

[Harinda]

Thanks for your reply, and it seem so easy to do that.

I tried with your setting , but when I try from another extension from the FreePBX , its only generate the incoming call.

May be something wrong with my SIPURA setting , would you mind posting your SIPURA setting so that I could give a try with them.

[Harinda]

I tried both of your method and various other methods from different forums , all I get was this response from Asterisk CLI. I did this testing continuous two days but so far no luck.

SIP/TEST-086b3130 is ringing
-- SIP/TEST-086b3130 answered SIP/2000-086c80a8
-- Packet2Packet bridging SIP/2000-086c80a8 and SIP/TEST-086b3130
Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2000-086c80a8' in macro 'dialout-trunk'

Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/2000-086c80a8'

Greatly appreciate someones generous advise .