On Tue, 7 Sep 2004 16:26:24 -0700, Kris Boutilier <[EMAIL PROTECTED]> wrote: > I'm having a problem with intersite calls over IAX2 being abruptly > terminated. Nothing odd shows in any of the logs for Asterisk or the host. > The only think I can think it might be is a lag-spike on the site to site > connection.
When does the cut off occurr? Is it always after about 8-10 seconds? If so, you may have a problem with IAX transfer. You can verify this by using notransfer=yes. > How sensitive is IAX2 to lost frames, lag spikes or large variations in > jitter with the GSM codec <snip> > > During an average call 'iax2 show channels' > provides: > > Peer Username ID (Lo/Rem) Seq (Tx/Rx) Lag Jitter > JitBuf Format > 10.0.40.140 astpbx-woo 00002/00002 00005/00006 00040ms 0036ms > 0000ms GSM Those values are certainly no problem for IAX at all. I have made hour long IAX calls with both lag and jitter often going well above 1 second and the calls never terminated. All you get is a heavy delay on the audio and occasional drop outs, but you shouldn't get cut off. Even if the lag goes above 2 seconds and you have qualify=yes, the calls will not normally be cut off when Asterisk reports "peer now to lagged". As long as the lag will go back below 2 seconds within a reasonable time frame the connection will recover. IAX is extremely robust, it is rare to have a connection terminate due to network problems. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
