Cisco SPA303 no audio after SIP reinvite

Asterisk Asterisk Support
1

Hi,

I’m testing cisco IP phone SPA303 against Asterisk 1.6.0.26.
I’m facing a strange problem that sometimes I can hear only initial part of the first prompt when making an outbound call from SPA303 to a landline through a SIP provider, but sometimes I can hear the entire prompt (about 20 seconds long).

According to wireshark, in those cases where I heard the whole prmopt, asterisk always stayed between the end-points for RTP traffic. And in other cases where I heard only the partial prompt, asterisk didn’t always stay between the end-points for RTP traffic and the problem happened when those both end-points tried to talk to each other directly.

Any idea what might be causing this problem? Thanks.

2

How does Asterisk decide to stay between end-points for RTP traffic?

3

canreinvite=no for either side (now renamed directmedia).
recording the call
DTMF operated features enabled for either side
one side isn’t SIP
Non-optimisable local channel in the circuit

There may also be NAT related criteria, but I’m not sure.

4

Still unable to fix it. The workaround is to force RTP traffic to always go through asterisk.