[Asterisk-Users] Asterisk 1.2 problems

jonc
Fri, 02 Dec 2005 14:01:59 -0000

On Fri, 2005-12-02 at 14:22, tneuwert@formos.com wrote:
> Help! I've encountered some problems with Asterisk that I?m unable to
solve. We have been running Asterisk version 1.0.9 for many months using a few
local network connected Cisco 7960 phones as SIP clients.  All our phones are
currently internal so there is no NAT involved.  We were not having any problems
until last week when some strange issues started to crop up. I started
experiencing calls that I initially believed were being dropped, but discovered
that only one side of the conversation had dropped.  The other party could hear
me but I couldn't hear them. This seems to happen more often on longer calls
but is not consistent.  I am also seeing issues where incoming or local
extension calls that are hung up by the originator before being answered will
continue to ring the SIP phone. At the time the errors occur, the Asterisk
console displays a variety of "...retrans_pkt: Maximum retries exceeded on
call.." messages. I scoured the forums for an answer, found many refere
 nce
>  s to these errors, tried every suggested fix that I could find, but none
have resolved these problems.  After working on the problem for several days, I
finally built a new box and installed Asterisk 1.2 on it. Using this new 1.2 box
I no longer see the "Maximum retries exceeded on call" warnings on the
console but still experience the strange behavior. Unfortunately, the errors
occur randomly so I am unable to reproduce the error on demand. I turned on SIP
debugging and set console logging to debug and captured an instance of the
problem with the hang up not being recognized.  The details are below:
>
> I dial in from my cell phone. My Cisco phone begins to ring. I then hang up
my cell phone. Asterisk acknowledges the hang up, but the Cisco phone continues
to ring. After a minute or so, or if I pickup the phone, Asterisk display the
following message "That's odd...  Got a response on a call we don?t
know about. Cseq 102 Cmd SIP/2.0"  I've included a copy of the console
output when this occurs that shows both the SIP message and the Asterisk debug
output.

Odds are you have local network congestion -- Dropped packets or delayed
packets.  Try moving your phone and asterisk server to an isolated
network switch - no other traffic (certainly no computers) - then test.

If the problems go away, then update your virus scanners and check your
computers.

Good Luck

Jon Carnes