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Author Topic: AA picks up and NumberOnNoInput is being called, but no sound on either side  (Read 3031 times)
perabino
Newbie
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Posts: 1


« on: October 22, 2015, 08:36:27 pm »

I've got an interesting situation...
  • A caller dials my intercom number.
  • Then, the Attendant does succeed in picking up the call.
  • After waiting for about 20 seconds, the Attendant dials the "NumberOnNoInput", which is my cell phone.
  • However, once I pick up my cell phone, I hear silence on the other end.
    • When I talk, the other end can't hear anything.
    • When the other end talks, I can't hear anything.
    • The local logs show that the call was transferred, and bridged too I believe (see attached log).


Some side notes that may/may not be related:
  • Call made from the phone attached to my Obi110 work fine (both sides can hear each other).
  • After the caller calls and the Attendant picks up, only the "main menu" user voice recording plays. In other words, the caller only hears the main menu recording as soon as the Attendant picks up, then from then on nothing until I hang up my cell phone- then the caller hears a dial tone.
  • I am just trying to forward intercom calls to my cell, and so I would just forward all the calls that come in through the Line port to SP1, but this type of intercom has 3 seconds of delay in between each ring (more than the max of 1920ms), so it keeps thinking that the original caller hung up the phone and ending the call before the next ring has a chance to start... that's why I'm using the auto attendant to pick up and then route the call to my cell.




* Call Log.png (68.48 KB, 762x388 - viewed 203 times.)
« Last Edit: October 22, 2015, 08:42:06 pm by perabino » Logged
ianobi
Hero Member & Beta Tester
*****
Posts: 1828


« Reply #1 on: October 24, 2015, 05:22:35 am »

The usual reasons for audio problems are RTP not being routed correctly or CODEC mismatches.

I have just simulated the problem you describe, but I don't have GV so I've used a test number at sip2sip as my "NumberOnNoInput". It worked fine. Try looking at Status > Call Status while your silent call is actually in progress. It should look something like the attached jpg. I've only posted the top half of the page as the part that matters concerns all the RTP and CODEC lines.


* AA_Bridged.jpg (172.65 KB, 1136x528 - viewed 194 times.)
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