On Sunday 23 January 2011 15:33:14 Stan Goodman wrote:
The system is openSuSE v11.2, running on a Dell Vostro laptop.
Today I tried to reproduce the experiment, this time dialing a US number: 1(201)555-1212. I attach the entire section of the log relating to the time at which I did this. I am not able to make much of the log, which I attach herewith. Any helpful insights, especially from users of Twinkle will be gratefully received.
For what it's worth, I have found that, although the logs seem to be saying that ALSA is in order (and the manual indicates that I would not be permitted to dial if there is no sound driver), the file /etc/asound.conf (mentioned in the Manual) does not exist. I do not understand this, because I am able to hear CDs. DVDs, and online sound streams.
I'm using 11.2 at work and 11.3 at home and on both I had to select the actual ALSA device from the device list instead of the 'ALSA default', but that may be because I've configured ALSA to use PulseAudio. Have you tested the sound configuration by making a direct SIP call in your local (home) network to another machine and checked that way that your audio works both way? Besides the audio configuration there is a ton of other stuff that might prevent you from hearing the other end, like e.g. firewall blocking the incoming RTP packets, using NAT, missing or incorrect SIP proxy and STUN server settings, etc. Looking at the log I can see you must be behind NAT since you are connecting from a non-routable address (192.168.x.x), so you should probably configure your SIP provider's STUN server address in your user profile, unless you already have done so, but then I would expect Twinkle to use the correct external IP address in SIP headers. After receiving 183 response (meaning the other end is requiring reliable provisional responses), Twinkle opens the audio devices and starts sending and receiving RTP packets immediately instead of just playing the default ringback tone from file until the other end answers or rejects the call. At this point you should be able hear audio from the other end, _if_ he is sending anything. Mind you, the person you tried to reach hasn't actually answered the call yet. And in this case he doesn't - the call is released with '403 Termination not possible' response, perhaps the person you tried to reach was not available? -- Martti -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org For additional commands, e-mail: opensuse+help@opensuse.org