I'm "playing" with a Rowetel IP04 Asterisk appliance running Asterisk 1.4.4 (yes, I know: very backlevel), and have encountered a strange (to me) problem with it's interactions with a sip softphone (Linphone, running on a wifi-enabled tablet).
I have a dial-plan that, in part, looks like [default] exten => #9,1,SayNumber(9) exten => #9,2,Goto(#9,1) From the two analog lines attached to the IP04 FXS ports, dialing #9 at the dialtone gets a voice that just repeats "Nine... nine... nine" ad infinitum. However, if I dial the same #9 from the SIP phone (that otherwise can dial any other extension under the same "default" context), the sip phone falls back to the "nothing dialed" state, as if I hadn't dialed anything (or, perhaps, as if it got an immediate, silent, hangup from the IP04). A look at the SIP phone's log shows something interesting: instead of "#9", the log shows that I've "dialed" "%239". If I add to the [default] context another extension [default] exten => #9,1,SayNumber(9) exten => #9,2,Goto(#9,1) exten => %239,1,SayNumber(9) exten => %239,2,Goto(%239,1) I can now "dial" "#9" (%239) on the sip phone and get the same audio back as if I dialed "#9" on either of the analog handsets. My question is: is there an option, set of options, or dialplan application, in Asterisk 1.4.4 that would cause the sip phone to use the "#9" extension, rather than the "%239" extension? I'm thinking of a sip.conf option, or perhaps some dialplan application that would permit me to replace the dialed string with it's URIDECODEd string. Any suggestions? Thanks -- Lew Pitcher "In Skills, We Trust" PGP public key available upon request
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