Outbound routing

I’m trying to understand something about how outbound calls are routed and need a little help. I’ve done my entire configuration via CLI and have gotten to the point where what I need works except outbound calls via PSTN. What I have so far is “exten => 9.,1,Dial(DAHDI/3/{EXTERN:1})” in my extensions.conf file. When I dial something like 95551212 instead of the number being dailed automatically via the PSTN port I get a PSTN dial tone where I can then dial again via my provider’s dial tone. Is there a way to make it where when I dial 95551212 the call is initiated via my PSTN and automatically dialed?

You have a typo: {EXTERN:1} replace with {EXTEN:1}

I don’t have the screen directly in front of me but I believe it’s entered correctly in the config, I typed that mostly from memory. Is it supposed to work the way it currently is or is what I’m trying to do possible?

The rule 9. means the digit 9 and any lenght of any digits dial to the number entered, so yes you rule its ok, only if you have another rule like 9XXXXXXX wich is more specific and it tries to another trunk.

Can you show us your cli output when you dial.

He also needs an _ for a pattern match, but without it, I wouldn’t think the . would match. In this context, please screen scrape, not re-key. If you are running purely on Linux, you can select, normally, then paste with middle mouse button. If you are working with a terminal emulator in any windowing environment, it should be even easier.

Right I always forget the basic things :frowning:

Right I always forget the basic things :frowning:

Thanks guys, finally got access to the machine. Here’s a copy/paste of my extension entry:

exten => _9.,1,Dial(DAHDI/1/{EXTEN:1},30)

And here’s the asterisk output

Connected to Asterisk currently running on asterisk (pid = 2772)
Verbosity was 3 and is now 7
== Using SIP RTP CoS mark 5
– Executing [92142971@phones:1] Dial(“SIP/6002-00000000”, “DAHDI/1/{EXTEN:1},30”) in new stack
– Called DAHDI/1/{EXTEN}
[2011-07-15 15:17:49] WARNING[3199]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)
[2011-07-15 15:17:50] WARNING[3199]: chan_dahdi.c:4831 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)
– DAHDI/1-1 answered SIP/6002-00000000

As stated above I get routed to my trunk provider’s dial tone via the FXO interface but have to dial again to actually complete a call.

Note to self, found a problem with not including the :1 to drop the leading 9 but now I get a “we’re sorry, your call did not go through” message.

You miss the $ its ${EXTEN:1}

Bah! just figured that out and was going to come back and post my results :unamused: Test call dialed and working. Thanks for the input.