Loss of outbound audio during a call (v1.2.3)

Asterisk Asterisk Support
1

Ive run into a problem where during a call (VOIP), all of a sudden the person I called cant hear me, but I can hear them no problem. It seems to do it randomly. Any ideas? Where should I start looking ? I sure could use the help . . .

Trixbox v1.2.3 (Asterisk 1.2.12.1)

[quote]Sip.conf
[general]
vmexten=*97
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
disallow=all
allow=ulaw
allow=alaw
;context = from-sip-external ; Send unknown SIP callers to this context
context = from-trunk ; allows anonymous sip callers
dtmfmode=rfc2833
callerid = Unknown
externip=69.XX.XX.XXX
localnet=192.168.0.0/255.255.255.0

Sip_additional.conf
;incoming trunk settings
[69.XX.XX.XXX]
type=user
host=69.XX.XX.XXX
fromdomain=69.XX.XX.XXX
dtmfmode=rfc2833
context=from-pstn

;outgoing trunk settings
[out trunk name]
username=69.XX.XX.XX
type=peer
host=69.XX.XXX.XXX
fromdomain=XX.XX.XX.XXX
dtmfmode=inband ;rfc2833 and info wasnt working with my provider so set to inband temporarily
disallow=all
allow=ulaw

;Extension used for test calls
[2000]
username=2000
type=friend
secret=2000
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
pickupgroup=1
nat=never
mailbox=2000@device
host=dynamic
dtmfmode=rfc2833
disallow=all
context=from-internal
canreinvite=no
callgroup=1
callerid=Ext 2000<2000>
allow=ulaw[/quote]

2

if it’s happening mid-call i would probably suspect the ITSP initially. ask them to run a debug at their end for your host and analyse the output for you.

in the past i’ve had situations where a firewall has been setup to allow inbound connections from the registrar and proxy server at the ITSP, but then mid-call the media is handed over to another server at their end, killing audio in one direction. opening up the firewall rules to the subnet of the ITSP sorted that out.

is there any kind of IDS running on your gateway that might object to lots of small packets from a single host ?

3

I too started having this problem recently, however I am using IAX2.

When making an outbound call, the audio is lost extremely frequently (e.g., 5 seconds out of ever 10 seconds of outgoing audio is lost). The incoming audio works fine and the problem only happens with outgoing calls (incoming calls work fine).

My strong suspicion is that the problem is with the outbound VOIP server, but I have no way to know for certain.

Any ideas would be appreciated.

Mark MacVicar