Outbound Prefix routing used as TAC not working - Help

Hello All,

I am ‘new’ to Asterisk, but not to telephony or SIP. I am setting up my Asterisk to use as a test line into our Ingress SBCs (to mirror our customer’s experience and issues, specifically with DTMF throughput and call completion). I have near a dozen ingress SBCs that I have connected with SIP trunks to my Asterisk as peer connections. I need the testers/users to be able to dial an 8+5 digit code to select the appropriate trunk to test calls out on.

For example to select our Beaverton SBC to test through a user would dial 823201 and then the 11 digit number they want to test to. I originally had it set up like:

exten => _8232011NxxNxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net), but this sent the entire dialed string to the SBC (which rejected it because it was over the digit length match).

Once I realized it was sending all the digits (the 823201 and the 11 digit TN), I set it up like this:

exten => _823201|1NxxNxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net)

However with that pipe in there, it now says that the extension 823201 is not in [LocalSets] and still fails the call…

Can someone PLEASE help me understand what is happening here, and how to be get the Trunk Access Codes to work as I need them to? What exactly am I doing wrong??

The error as I received it from Asterisk:

[Mar 14 07:23:41] NOTICE[2016]: chan_sip.c:23272 handle_request_invite: Call from ‘Mike’ (10.xxx.xx.xxx:15804) to extension ‘82320113038983192’ rejected because extension not found in context ‘LocalSets’.

The ‘extensions.conf’ configuration for these specific endpoint:
exten => 1003,1,Dial(SIP/Mike)
include => external

exten => _823201|1NxxNxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net)
exten => _823201|Nxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net)
exten => _823201|0xxxxxxxxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net)
exten => _823201|NxxNxxxxxx,1,Dial(SIP/${EXTEN}@sbc01.company.name.net)[/b]

Where did you get the pipe notation from?

The sample configuration file includes examples of sub-stringing dialed numbers to remove access codes.

Many people here do not have session border controllers, but still understand such basic dialplan logic, so SBC will probably have confused them.

I googled and researched, and it was the only annotation I could find… (the pipe I mean)

Sample configuration file? what is the file name in Asterisk?

Thanks, and have a great day!


I do not have this file on my Asterisk… someone may have removed it? I just searched and cannot find an unedited version on the web… any suggestions?

Thanks again for your help so far… I appreciate it.

Replace YOURVERSION by the version number of your Asterisk in:

svn.digium.com/svn/asterisk/tags … onf.sample

Then complain to whoever supplied your Asterisk that they have missed out the primary documentation for the configuration files.

David55, thank you very much!! I will do so!