On 27/09/13 14:36, Joshua Colp wrote:
Gareth Blades wrote:
We have an issue with a customer where when calls are sent to one of
their offices as soon as the call is answered the call fails.
We are performing remote bridging and switching the audio from the
server which initiated the call to our switch which is on the same
network.
After the call is answered we switch the audio which is accepted fine
but we then send the following packet and get a SIP/488 response from
the far end.
This packet seems to be updating the version for the o= session id which
is fair enough. Its sending the c= connection data but not the
m=audio line
which appears to be what the remote end is complaining about.
Can anyone with a bit more knowledge about the SDP standard tell me if
what asterisk is doing is correct?
Or if it must be a bug with our customers equipment?
The SDP you posted should be fine BUT my question becomes... have you
modified chan_sip at all? I don't think it should be possible for it
to not put any media lines in...
Cheers,
No we havent made any changes to chan_sip. The servers were a fresh
install a short while ago straight to 11.2-cert1 as we wanted a later
kernel version to make use of the new timing source it provides. We then
upgraded to cert2 after it was released.
The only thing we have changed is the setting of a DYNAMIC_FEATURES
variable which was stopping remote bridging from being performed which
is probably what has highlighted this fault.
Thanks
Gareth
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