Hello everyone. I am having trouble receiving via my Linksys SPA-3102. I have not problem dialing out. It is like asterisk never even sees the call. I have 3 sip devices grandstream bt-100, spa-3102 fxs, and spa-3102 fxo. A very simple setup, just getting familar with asterisk. Here are my relative config files. let me know if you need more. sip.conf [general] context=default bind=0.0.0.0 bindport=5060 srvlookup=yes [100] ;bt-100 type=friend username=100 context=default secret=secret host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=100@default [101] ;fxs type=friend username=pots context=default secret=phone host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw mailbox=101@default [102] ;fxo type=friend context=default secret=pstn host=dynamic dtmfmode=rfc2833 disallow=all allow=ulaw port=5061 extensions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no context=default [globals] RINGGROUP1 => SIP/100&SIP/101 [default] ; These next three lines are for testing, just to make sure I got the call, but no good exten => s,1,Answer exten => s,2,System(touch $HOME/got_it) exten => s,3,Hangup ;exten => s,1,Dial(SIP/100,10) ;exten => s,2,Hangup exten => 97,1,Dial(${RINGGROUP1},10) exten => 97,n,Hangup exten => 98,1,Answer exten => 98,n,AGI(agi-test.agi) exten => 98,n,Hangup exten => 99,1,Answer exten => 99,n,Playback(hello-world) exten => 99,n,Hangup exten => 100,1,Answer exten => 100,n,Dial(SIP/100,15) exten => 100,n,VoiceMail(100@default) exten => 100,n,Playback(vm-goodbye) exten => 100,n,Hangup exten => 101,1,Answer exten => 101,n,Dial(SIP/101) exten => 101,n,Hangup exten => _XXXXXXXXXX,1,Dial(SIP/102/${EXTEN}) exten => _XXXXXXXXXX,n,Hangup I appreciate your help - Jim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070128/14592166/attachment.htm
What appears on the Asterisk console? PaulH On Sun, 2007-01-28 at 20:06 -0500, James Caffrey wrote:> Hello everyone. I am having trouble receiving via my Linksys SPA-3102. > I have not problem dialing out. It is like asterisk never even sees > the call. I have 3 sip devices grandstream bt-100, spa-3102 fxs, and > spa-3102 fxo. A very simple setup, just getting familar with asterisk. > Here are my relative config files. let me know if you need more. > > sip.conf > [general] > context=default > bind=0.0.0.0 > bindport=5060 > srvlookup=yes > > [100] ;bt-100 > type=friend > username=100 > context=default > secret=secret > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > mailbox=100@default > > [101] ;fxs > type=friend > username=pots > context=default > secret=phone > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > mailbox=101@default > > [102] ;fxo > type=friend > context=default > secret=pstn > host=dynamic > dtmfmode=rfc2833 > disallow=all > allow=ulaw > port=5061 > > extensions.conf > [general] > static=yes > writeprotect=no > autofallthrough=yes > clearglobalvars=no > context=default > > [globals] > RINGGROUP1 => SIP/100&SIP/101 > > [default] > ; These next three lines are for testing, just to make sure I got the > call, but no good > exten => s,1,Answer > exten => s,2,System(touch $HOME/got_it) > exten => s,3,Hangup > ;exten => s,1,Dial(SIP/100,10) > ;exten => s,2,Hangup > exten => 97,1,Dial(${RINGGROUP1},10) > exten => 97,n,Hangup > exten => 98,1,Answer > exten => 98,n,AGI(agi-test.agi) > exten => 98,n,Hangup > exten => 99,1,Answer > exten => 99,n,Playback(hello-world) > exten => 99,n,Hangup > exten => 100,1,Answer > exten => 100,n,Dial(SIP/100,15) > exten => 100,n,VoiceMail(100@default) > exten => 100,n,Playback(vm-goodbye) > exten => 100,n,Hangup > exten => 101,1,Answer > exten => 101,n,Dial(SIP/101) > exten => 101,n,Hangup > exten => _XXXXXXXXXX,1,Dial(SIP/102/${EXTEN}) > exten => _XXXXXXXXXX,n,Hangup > > I appreciate your help > > - Jim > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
nothing On 1/28/07, Paul Hales <pdhales@optusnet.com.au> wrote:> > > What appears on the Asterisk console? > > PaulH > > On Sun, 2007-01-28 at 20:06 -0500, James Caffrey wrote: > > Hello everyone. I am having trouble receiving via my Linksys SPA-3102. > > I have not problem dialing out. It is like asterisk never even sees > > the call. I have 3 sip devices grandstream bt-100, spa-3102 fxs, and > > spa-3102 fxo. A very simple setup, just getting familar with asterisk. > > Here are my relative config files. let me know if you need more. > > > > sip.conf > > [general] > > context=default > > bind=0.0.0.0 > > bindport=5060 > > srvlookup=yes > > > > [100] ;bt-100 > > type=friend > > username=100 > > context=default > > secret=secret > > host=dynamic > > dtmfmode=rfc2833 > > disallow=all > > allow=ulaw > > mailbox=100@default > > > > [101] ;fxs > > type=friend > > username=pots > > context=default > > secret=phone > > host=dynamic > > dtmfmode=rfc2833 > > disallow=all > > allow=ulaw > > mailbox=101@default > > > > [102] ;fxo > > type=friend > > context=default > > secret=pstn > > host=dynamic > > dtmfmode=rfc2833 > > disallow=all > > allow=ulaw > > port=5061 > > > > extensions.conf > > [general] > > static=yes > > writeprotect=no > > autofallthrough=yes > > clearglobalvars=no > > context=default > > > > [globals] > > RINGGROUP1 => SIP/100&SIP/101 > > > > [default] > > ; These next three lines are for testing, just to make sure I got the > > call, but no good > > exten => s,1,Answer > > exten => s,2,System(touch $HOME/got_it) > > exten => s,3,Hangup > > ;exten => s,1,Dial(SIP/100,10) > > ;exten => s,2,Hangup > > exten => 97,1,Dial(${RINGGROUP1},10) > > exten => 97,n,Hangup > > exten => 98,1,Answer > > exten => 98,n,AGI(agi-test.agi) > > exten => 98,n,Hangup > > exten => 99,1,Answer > > exten => 99,n,Playback(hello-world) > > exten => 99,n,Hangup > > exten => 100,1,Answer > > exten => 100,n,Dial(SIP/100,15) > > exten => 100,n,VoiceMail(100@default) > > exten => 100,n,Playback(vm-goodbye) > > exten => 100,n,Hangup > > exten => 101,1,Answer > > exten => 101,n,Dial(SIP/101) > > exten => 101,n,Hangup > > exten => _XXXXXXXXXX,1,Dial(SIP/102/${EXTEN}) > > exten => _XXXXXXXXXX,n,Hangup > > > > I appreciate your help > > > > - Jim > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070130/9604bb64/attachment.htm