Diff between autocreatepeer and manually entering exten?

Asterisk Asterisk Support
1

Hi,

When I use autocreatepeer I get one way audio from pots back to sip. The SIP phone can hear the POTS phone, but not the reverse. Since I am new to this, I am still trying to figure out what I am missing. I am inside on a LAN, no NAT’ing involved. PBX and SIP phone are on same subnet.

Below, is a snippet from my sip.conf. If I use in this manner, I don’t have the 2 way audio problem. If I use autocreate peer, I only get one way audio. What’s missing?

As an aside, I can’t find any docs on autocreatepeer. Are there other features to this option? For example, if I want to create 20 phones, can I use autocreatepeer, but ensure the user configures a username and secret? Thanks!

[82010]
type=friend
context=incoming
;username=chicago
;secret=snom
host=dynamic
dtmfmode=auto
canreinvite=no
nat=no

2

NOTE for my above message. I am not using any special HW in the * box. My Snom phones register with the *, and I bring calls in via my Cisco Gateway to the * server. The call originates on our main building PBX.

3

figured out my solution. For autocreatepeer, I set canreinvite=no, and walla, I have 2 way RTP.

My implementation is not a standard pbx. I am trying to sign a user in to a phone, then, once signed in, send him a call and get two way audio. All is working the correct way now.

If I was just doing plain old pbx, which I should have mentioned yesterday, I don’t have a problem. If I send a call to a SIP phone from POTS, I don’t have audio problems. It’s my special requirements to sign someone in that causes me to have to use canreinvite - if you care to know.