No subject


Thu Jul 12 09:23:04 CDT 2007


of date and/or wrong, is that auto congest used to be calculated as qualify
*2, which was later changed to qualify*4 (see
http://bugs.digium.com/view.php?id=765  )

This I understand was later changed again to use timerb in sip.conf which
defaults to 64*timert1 = 32 seconds. Changing these values does not seem to
effect peers that do not register however and reading voip-info we get:
'For instance, a 1.6 version of Asterisk would not use the <snip> 'qualify'
<snip> columns, ...' (see
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip   ) 

I’ve included the providers SIP information attached as siptable.txt,
there is no registration, they just accept calls across a vpn link to an IP
address, if they are down we want to send it out to a different provider. I
have tried setting the qualify value in ms as well as just set to yes with
no difference.

I have tested it by sending both numbers that it will not recognise and by
modifying the ip so it is incorrect. In the first scenario it works like a
charm as we get sip error 500 and it flows through the next priority,
however in the later it takes the full 32 seconds before it fails over
(which is too long for an end user) and we would like to set this to
something more reasonable like 3 or 5 seconds.

The settings I have set in sip.conf have been attached as sipconf.txt

Let me know if you need anything else.

All the best - Kactus 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-29-08 21:10  kactus         Note Added: 0086169                          
======================================================================




More information about the asterisk-bugs mailing list