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) 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 Philip Prindeville wrote: > I've got the following situation. I'm running Asterisk 1.4.18 on a > firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones > behind it. > > I'm peering SIP with a Coppercom switch sitting behind an SBC. > > On outbound calls, I get 2-way voice, no worries. > > On inbound calls, I get one-way voice (I can hear the caller but they > can't hear me). > > I've looked at tcpdumps of the RTP traffic, and the addresses and port > numbers correspond to what's in the SIP INVITE/OK messages (assuming > that they don't somehow get munged by NAT after tcpdump looks at them -- > there is no NAT device upstream of my Asterisk firewall). > > I'll look into using Record() or Monitor() to capture the phone call, > but if there's any conversion being done by codecs then that won't > eliminate the possibility that the code itself is misconfigured or buggy > and generating a bad stream on one of the legs... > > Anyone have an idea about how to best go about troubleshooting this? > > 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