Hi folks - hope you can help with this one…
The Asterisk server in one of my offices is experiencing bad audio problems - when they call out to anyone - the audio starts to sound broken up - garbled - like talking underwater. They are running exactly the same version I am 1.4.24.1 the only difference I see is in their configs which I did not set up since I inherited these and were in place.
I don’t know where to look so I am starting with the config files. Their Extensions.conf file has this context…
[bandwidth]
exten => _+1.,1,Set(DID=${EXTEN:2})
exten => _+1.,n,Goto(fromtrunk,${DID},1)
and the sip.conf file shows …
[bandwidthPri]
host=216.82.224.202
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
reinvite=no
canreinvite=no
context=bandwidth
nat=no
[bandwidthAlt]
host=216.82.225.202
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
reinvite=no
canreinvite=no
context=bandwidth
nat=no
[bandwidthOut]
host=216.82.225.202
port=5060
type=peer
allow=ulaw
dtmfmode=rfc2833
nat=no
reinvite=no
canreinvite=no
Also - when a call is made from their system I see i.e. 8005551212@bandwidthout in the CLI as I watch to see what is happening.
None of my other systems are configured to show "BandwidthOUT - ALT or PRI in any config file.
I believe the IP address shown is the carrier that office is using ( bandwidth.com )
I am trying to isolate the sound issue and this is the only difference I see on their system.
I don’t even know if this is where to look but figured it was a start.
I do hope you can help,
Thank you.