Pause Before Answer Cancels audio?

Hi, I have a strange thing happening, I am a novice so please bare with me.

Everything is working great on my setup, but one of the bosses is complaining that they “Want the caller to hear the phone ringing, because it gives a more personal feel”, where as it stands right now our Sip trunk answers prior to it ever ringing and starts playing our welcome message.

SO I thought this was an easy thing, but when I modify the inbound route to Signal Ringing and pause before answer to 6 seconds, it now RINGS for 6 seconds and I hear it, but then NO AUDIO can be heard by the caller. The call is routed correctly but the caller can’t hear a thing.

I am not sure if this is a codec issue as I was messing around with them earlier today, but I don’t think so since it works perfectly find when I remove those settings.

I am not sure what changes by enabling them, but anyone know where to start?

This doesn’t seem to be a question about Asterisk. It might be about an Asterisk GUI.

If it is really about Asterisk, you need to provide the dial plan code that is being executed.

Here is the Answer plan, it works, just no audio comes through. If I comment out ringing() and wait(6) no issue.

include => ext-did-0002-custom
exten => 16302063075,1,Set(__FROM_DID=${EXTEN})
exten => 16302063075,n,Gosub(app-blacklist-check,s,1())
exten => 16302063075,n,Set(CDR(did)=${FROM_DID})
exten => 16302063075,n,ExecIf($[ “${CALLERID(name)}” = “” ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 16302063075,n,Ringing()
exten => 16302063075,n,Wait(6)
exten => 16302063075,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 16302063075,n,Set(CALLERPRES()=allowed_not_screened)
exten => 16302063075,n(dest-ext),Goto(app-announcement-1,s,1)

Are you sure you ever answer? This feels like an “early media” handling problem.

Otherwise, you need at least the verbose logs and may need the a SIP protocol trace.

In some circumstances, once you are commited to ringing, you may not have the option of switching to early media. That’s more a function of the upstream system.

I am sure it is answering and doing everything properly because the internal queues and lines ring, just no audio on the callers side.

I restarted my linux and it tends to be working now. My thought is I installed an open source G729 Codec and made that the ONLY active codec for testing. There may be an issue with it loading properly, I have a core 2 duo processor running in a emulator and the binary for that codec was a core 2 (not a duo), when it worked after the restart I am leaning towards it may be an issue with that codec. Unfortunately I can’t get the source to compile properly so I am stuck using the pre-compiled binary.

Ringing does not, in any way, indicate that the call is answered.

There aren’t any open source G.729 codecs, as far as I know. The only codec that I am aware of that purports to be open source has a void GPL licence, because it is a derivative work of code with a no commercial use restriction.

While testing it i also setup a DID to my extension, and even though i heard nothing, my vm got a message where i could hear it was transmitting, just no audio back to the caller. It hasn’t happened again, but whatever was causing it seemed to be remedied with a reboot. The g729 open source is the one you mentioned. Our system is still being setup. It is not live. We are going with digium phones and they will be providing our codec licensing as well.

Since you seem familiar with Asterisk, what do you think of the digium phones as far as quality? We are going with those over polycom, which was our original quote.