[Asterisk-Users] Asterisk 1.0 -- Did the SIP dial syntax change?

[Benjamin on Asterisk Mailing Lists]

I have successfully tested a configuration for dialling out through a
SIP based FXO gw on an older version of Asterisk and now that I moved
it to Asterisk 1.0, it works no longer.

Basically, I have defined a SIP peer in sip.conf called "fxogw" and I
dial a PSTN number like so ...

exten _9X.,1,NoOp(Outgoing PSTN call to ${EXTEN:1})
exten _9X.,2,Dial(SIP/${EXTEN:1}@fxogw,60,r)
exten _9X.,3,Hangup

This translated into dialling sip:pstn-number at gw-ip-addr:port

However, with Asterisk 1.0, this translates into sip:fxogw at gw-ip-addr:port

I also tried Dial(SIP/fxogw/${EXTEN:1},60,r) but that, too translates
into sip:fxogw at gw-ip-addr:port

So, now I wonder, where did the actual number dialled go? Did the
syntax of SIP dialling change?

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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