Philip Prindeville wrote: > Well, things just got a lot more interesting... Adding Monitor() to an > extension ends the one-way voice problem on inbound calls! > > So an incoming call gets handled as: > > [ctc-incoming] > exten => 208345****,1,Noop() > exten => 208345****,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI: > ${CALLERID(ani)}) > exten => 208345****,n,Goto(redfish-pstn,s,1) > ... > > [redfish-pstn] > exten => s,1(incoming),Noop() > exten => s,n,Answer() > exten => s,n,Wait(0.5) > ... > some filters for bogus ANI's like 888888888.... goes to badani below > > exten => s,n(exten),Background(vm-enter-num-to-call) > exten => s,nWaitExten(5) > exten => s,n(goodbye),Playback(vm-goodbye) > exten => s,n(end),Hangup() > > exten => s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing) > exten => s,n,Playback(privacy-unident) > exten => s,n,Wait(0.5) > exten => s,n,Congestion() > exten => s,n,Hangup() > > include => redfish-extens > > exten => i,1,NoOp(Invalid: ${EXTEN}) > exten => i,n,Playback(pbx-invalid) > exten => i,n,Goto(s,exten) > > exten => t,1,Goto(s,goodbye) > > [redfish-extens] > ... > > exten => 113,1,Monitor(wav,,w) ; for debugging > exten => 113,n,Macro(stdexten,113,${GUEST},redfish) > exten => 113,n,Goto(s,exten) > > ... > > exten => 113,1,Macro(stdexten,119,${GUEST},redfish) > exten => 113,n,Goto(s,exten) >
Err, sorry. Typo. That was: exten => 119,1,Macro(stdexten,119,${GUEST},redfish) exten => 119,n,Goto(s,exten) -Philip > So I don't get this at all. If I dial 208345****, then enter '119' as > the extension, it rings on a few phones (including a Xlite softphone) > and if I pick up on any of those, I get one-way voice (I can hear the > caller but they can't hear me). > > If I enter '113' as the extension, it rings on two SPA-942's (one of > which is the same as above, just a different line presentation)... and > if I answer, then I get two-way voice! Only difference is the Monitor() > statement. > > I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why > Asterisk would need to transcode a call between two uLaw endpoints, I > don't know... and (b) why is it staying in the Media path at all? > > I have the SIP peer that the calls come in on as: > > [sip-proxy] > ... > type=peer > nat=no > canreinvite=no > reinvite=no > > Anyone know why the Monitor() would change the duplex(ity) of the audio > stream? I'm baffled (no pun intended). And is there any debugging I > can turn on to reveal CODEC behavior that might differ from 113 and 119? > > Thanks, > > -Philip > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users