[Asterisk-Users] Sip One way audio

[Paul Oster]

I've got a telecommuter working out of her home office, using a Snom 200
phone, what happens is occassionally her phone will loose audio one way.

She will be talking on a call that was incoming to her extension, and all of
a sudden the caller can not hear her any longer, she can here the caller
fine, this has happened with both Sip->Sip calls, and calls that have come
in over our PSTN circuits.  The really odd thing is while troubleshooting
with her yesterday I was using the one way audio to talk to her and do some
packet captures, and she was using an instant message client to communicate
back to me, but after being in the call for a while (didn't note exact
times) the audio came back.

At first I thought this was a nat issue, and she is using Bellsouth DSL, so
I had her change the dsl modem so it shares its IP address with the phone.
Restarting the phone results in the phone getting the public IP address
assigned via DHCP.  This did not solve the issue.  I've experimented with
the nat settings, and the canreinvite settings but haven't had much sucess
so far.  I have suspicions that the cut-outs might be occuring either after
the phone has been registered for a certain amount of time (possibly 1 hour)
or when she has been talking for a certain amount of time (possibly 5
minutes), I'm not certain of that behavior so it may be a red herring
further use of the phone will allow me to firm up if either of those
statements is true.

Any suggestions would be greatly appreciated!

Thank You

Paul M. Oster


Here are the relevant portions of my sip.conf file...


[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = incomingcall          ; Default for incoming calls
tos=lowdelay
disallow=all
allow=alaw
allow=gsm
allow=ulaw


[104]
accountcode=vsllc
type=friend
context=employee
username=104
secret=**redacted**
host=dynamic
qualify=yes
reinvite=no
canreinvite=no
mailbox=104 at internalextensions,750 at internalextensions
callgroup=1
pickupgroup=1
dtmfmode=rfc2833
;nat=no
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060209/d29e2a97/attachment.htm