no audio on sip devices

[jweek]

Hi -

I've been searching, but not able to find any answer to my problem. I'm running Piaf, latest scripts, fixes and source code as a virtual machine on server 2008 hyper-v. Alll my sip devices are on the same subnet, no nat. I have a grandstream bt101, linksys spa941, a couple of laptops running the latest version of zoiper and a spa3102. I also have an account with voipjet and have it configured as an outbound trunk for long distance.

All devices can place calls with normal audio via the voipjet (iax2) trunk. All sip devices can call each other, but after they connect, there is no audio in either direction. If I place one endpoint on hold, the moh plays fine on the other endpoint. After the endpoint is taken off hold, the audio works both ways.

I'm an asterisk rookie, but not a newbie. I had a stand alone version running for a couple years on another box and never had this problem.

Any ideas on where to look next?

thanks!

John

 

[jroper]

Hi

I'd be looking at each phone, rather than at the Asterisk Installation to begin with.

Dial *43, so you get to hear yourself on the echo test.

Check all the codecs, just set them all to ulaw to start with.

Other than that, you've got a strange problem.

Joe

 

[jweek]

Hi

I'd be looking at each phone, rather than at the Asterisk Installation to begin with.

Dial *43, so you get to hear yourself on the echo test.

Check all the codecs, just set them all to ulaw to start with.

Other than that, you've got a strange problem.

Joe

Joe -

thanks for the tip, found a couple of things:

the spa941, bt101 and spa3102 work perfect on the echo test, however, the laptops do not echo (I can hear the asterisk lady give the "echo test stuff" first). If I use a laptop to call one of the hardphones, no audio either way at first, then after putting the call on hold (moh plays fine), audio is passed one way from the hardphone to the laptop. everything is set for ulaw.

John