Silence after upgrade to 1.2.2 (bug) v1.2.3 now available


#1

Hello,

We upgrading to asterisk 1.2.2 yesterday and made the changes listed in the UPGRADE.txt file. After the upgrade evething worked fine but this morning all we get is silence we dial internally or externally.

I suspect that phones registeration expired somehow and the phones are now using different settings but I cannot see what.

No errors can be seen on the CLI (SIP debug) or on the sip logs on the phones. We are using SNOM 360’s with various firmware 4 to 5.2.

Any ideas welcome If I dont get this working soon will have to downgrade to * 1.0 which I am not keen to do.

Many Thanks…

MP//


#2

upgrade to 1.2.3 there was a bug in 1.2.2 when it reach january 25th 2006, so change the date of the server to before january 25th or upgrade to 1.2.3. That fix the problem for me.


#3

Thanks mate that saved me a lot of debugging time :smile: Had the same issue.


#4

Fixed! Thanks very much.


#5

Two sites down this AM on this, arrrgh! Of course I struggled and found messages pointing out the bug only some time later.

Maybe this could have been on Asterisk-Announce or some other alert list? It was on Asterisk-Users, but I don’t subscribe because of the load.

Suggestions, anyone? This is important info to get out.

Thanks for the heads up, Andy


#6

Yes, I had the same problem. Keep this as a sticky for a little bit. Critical update details here:

asterisk.org/asterisk-1.2.3


#7

running ‘make update’ fixed it for me too (after a very frustrating day!)

Surely this should have been announced.


#8

Well, I managed to completely b0rk my AAH 2.2 system. Apparently updating to the latest asterisk via subversion was NOT the right thing to do. Ugh…


#9

I looked at what AAH 2.3 had for source changes, and sadly, it did NOT seem to include the time bomb patch in channel.c :frowning: I manually patched it and here it is:

— channel.c.orig 2006-01-10 23:38:07.000000000 -0500
+++ channel.c 2006-01-26 23:19:06.000000000 -0500
@@ -3233,11 +3233,14 @@
res = AST_BRIDGE_RETRY;
break;
}

  •            if(bridge_end.tv_sec) {
               to = ast_tvdiff_ms(bridge_end, ast_tvnow());
               if (to <= 0) {
                       res = AST_BRIDGE_RETRY;
                       break;
               }
    
  •            } else
    
  •                  to = -1;
               who = ast_waitfor_n(cs, 2, &to);
               if (!who) {
                       ast_log(LOG_DEBUG, "Nobody there, continuing...\n");