[Asterisk-Users] Asterisk 1.2 problems

[tneuwert at formos.com]

> On Fri, 2005-12-02 at 14:22, tneuwert at formos.com wrote:
>> Help! I've encountered some problems with Asterisk that IÂ’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 donÂ’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
>

Thanks for the feedback.  We will definitely try this and let you know how it goes.  Do you know if there is a way for us to observe the loss of packets?  Because of the sporadic nature of the problem, we have thought several times that the problem was solved only to have it crop back up hours later.  If we ran Ethereal on the pbx, would we be able to see that packets were being dropped?

Thanks,
Tim