There might also be a way to configure the SIP phone itself to use a
more correct format for the invite, thus eliminating the need to adjust
your dialplan.
--James
On 06/27/2013 12:30 PM, Shamus Rask wrote:
This isn't AstLinux specific, however I can no longer find any
helpful, active Asterisk forums. The digium ones just seem dead...
I'm currently running the latest version of AstLinux with Asterisk
v1.8.21. I have an existing dialplan that allows for basic NANP
dialing in the form of:
exten => _1NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},100,KT)
and
exten => _NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},100,KT)
I've recently purchased a SIP phone that pulls my Gmail contacts into
its address book. These contacts have their phone numbers stored as
(e.g.) (716) 452-1234 <tel://%28716%29%20452-1234> (note parenthesis).
This was not working with the existing extens listed above, so I added
the following:
exten => _(NXX)!NXX!XXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN},100,KT)
This seems to work as expected (i.e. for example above it dials
7164521234 <tel://7164521234>). Is there a better/safer way to do
this? Does anyone have any better suggestions?
cheers,
Shamus
--
Shamus Rask
sip:sha...@srask.ca
ISN:100*1129
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