Re: [GnomeMeeting-devel-list] SIP, RTP ports and firewall

Damien Sandras a écrit :

Le dimanche 20 novembre 2005 à 22:51 +0100, Daniel Huhardeaux a écrit :
Damien Sandras a écrit :

Daniel Huhardeaux a écrit :

Daniel Huhardeaux a écrit :

Damien Sandras a écrit :


If the snapshot you are using is correct, the key name is :

Ok got it. Thanks


Please modify RTP port range to 16384:16388. Doing this, when I place
a call it never stop to ring I have to kill GM :-(

Some else see this?

I think there is something wrong with this: I put 6970:6990 for RTP
port. I can now pass 2 calls and have no audio on the third. Does this
have a sense? FYI my asterisk uses 6991:7170

on the LAN?


Then this doesn't make sense =)

Can you debug further? (Please upgrade first).
Upgrade done: if RTP ports stays to 6970:6990, same behaviour. Two calls, that's all.

If I modify and put 16384:16388 I now get a Error Popup window: Erreur Générique - Function pthread_mutex_destroy failed. Push validated, GM hangs up properly with an "abnormal end of call"

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