Occasional One-Way and Calls Failing

General Help
1

Hello all,

I am having trouble with my FreePBX. When I can ever manage to make calls between a phone registered under my Asterisk and another phone, the outside line has limited audio. The outside line will not send audio in regular intervals, usually allowing for 6-8 seconds of clear audio and dead silence for about 30 seconds.

On top of this, calls will not always make successful calls either. I can make calls occasionally, but it has a low success rate. I have checked several posts about this and they do not seem to remedy my problem, I believe it is related to the first problem. On the web interface, the IP Trunk Registrations bar occasionally blinks off.

Here’s a list of my Asterisk box credentials:

FreePBX Version: 2.3.1.7

Trunk Configuration:

Trunk Name: trunk-out
PEER Details:
allow=ulaw&alaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=unlimitel.ca
host=sip02.unlimitel.ca
insecure=invite
progressinband=no
relaxdtmf=yes
rfc2833compensate=yes
secret=secret
type=peer
username=6133694437

User Context: trunk-in
allow=ulaw&alaw
canreinvite=no
context=from-pstn
disallow=all
dtmfmode=rfc2833
fromdomain=unlimitel.ca
host=sip02.unlimitel.ca
insecure=invite
progressinband=no
relaxdtmf=yes
rfc2833compensate=yes
secret=secret
type=peer
username=613NNNNNNN

Register String:
613NNNNNNN:[email protected]/613NNNNNNN

General Settings:
Allow Anonymous Inbound SIP Calls?

Inbound Routes:
any DID / any CID

Edit: Remove sip debug text

2

Man that is a lot of nothing.

search for sip_nat.conf

search for RTP ports

3

I have come across another problem. I have added another phone to my FreePBX and if I try calling each other through the network, I get clear and quick call. If I use one phone and call out and call back in, I get no audio at all.

4

see Bubba’s posting above.

At the same time you did provide some info but not enough of the right stuff.

See: http://freepbx.org/forum/freepbx/installation/so-you-have-a-problem-and-want-help

5

I agree with Bubba; this is probably a NAT problem that can be fixed with the appropriate modification to sip_nat.conf. One of the things that I noticed from the above configs however is the type is set to peer in both the inbound and outbound settings. It needs to be peer in the outbound and user in the inbound.

Outbound

type=peer

inbound

type=user