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?
U 2013/09/27 11:04:55.352854 88.x.x.25:5060 -> 213.x.x.24:5060
INVITE sip:firstname.lastname@example.org:54900 SIP/2.0. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713. Route:<213.x.x.24>