vicidial.org

by **mflorell** » Fri Sep 08, 2006 4:45 pm

I have confirmed today a bug in Asterisk 1.2.11 involving meetme and SIP or IAX trunks. What happens is in auto-dial mode or on inbound calls, it sounds like a call has been sent to an agent, but there is no audio.

If you are using SIP or IAX trunks please use Asterisk 1.2.10 instead.

Here's a link to the bug on Digium bugs:
http://bugs.digium.com/view.php?id=7909

Please monitor this bug so that we can get more attention on this.

by **smt8d1** » Fri Sep 08, 2006 6:33 pm

I was banging my head against a wall trying to figure out what I was doing wrong. =)

by **espencer** » Fri Sep 08, 2006 10:12 pm

holy moly -- my symptoms exactly!!

nice find!

by **espencer** » Sat Sep 09, 2006 12:50 pm

asterisk 1.2.12 released just now:

http://www.asterisk.org/asterisk-1.2.12

does any of this sound like it fixes the problem we are experiencing:

Asterisk 1.2.12 includes a number of bug fixes, including fixes for two regressions that occurred in the 1.2.11 release. Specifically, the AGI 'GET VARIABLE' command has now gone back to its previous behavior, and CDR records now reflect the CallerID number instead of ANI in the situations that this was the case in earlier 1.2 releases.

by **mflorell** » Sat Sep 09, 2006 3:06 pm

I will test it on Monday, but I don't think that's it, I believe the problem is in the removal of a few routines that did transcoding in the channel.c code.

by **gerski** » Sat Sep 09, 2006 11:40 pm

Hello mflorell,

That is my problem on one of my server running channel bank sending iax channels. the CLI output said the call has been sent but nothing happen on agents.. :cry:

im using app_conference now but the load average of my server is around 6.0 to 10.0

Hope we can fix this

by **mflorell** » Sun Sep 10, 2006 12:48 am

This bug has nothing to do with load. And with a load of 6.0 to 10.0 you will have many problems. You need to distribute the load across more computers.

by **gerski** » Sun Sep 10, 2006 1:15 am

what i mean is that meetme conference does not work wit me because of no audio is sending to the zap.

right now im using app_conference and i think it loads more processor than meetme, am i right?

by **mflorell** » Sun Sep 10, 2006 1:37 am

app_conference almost always uses less CPU than meetme. But it is not as stable.

by **gerski** » Sun Sep 10, 2006 1:38 am

ok will review my installation again.. do you have any suggestion as of now to use meetme? do u suggest to install lower version of asterisk?

by **mflorell** » Sun Sep 10, 2006 2:59 am

Yes, use Asterisk 1.2.10

by **mflorell** » Mon Sep 11, 2006 11:53 am

Bug is fixed in Asterisk SVN asterisk 1.2

I have also posted a patch for Asterisk 1.2.12:

(asterisk 1.2.12) - channel.c bug fix -
**THERE IS NO FIX FOR ASTERISK 1.2.11 DO NOT USE IT**
wget http://www.eflo.net/files/channel.c-42600.patch
patch -p1 < ./channel.c-42600.patch
- File to patch: channel.c

by **Op3r** » Mon Sep 11, 2006 12:11 pm

Thats the reason why we still use 1.2.10

we are always being carefull when we install asterisk thats why we dont install the latest version

by **gerski** » Mon Sep 11, 2006 10:01 pm

we are now using Asterisk SVN asterisk 1.2 and it works perfectly

vicidial.org

Bug in Asterisk 1.2.11

Bug in Asterisk 1.2.11

Thanks for posting this

holy moly

asterisk 1.2.12

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