asterisk users - Sep 2003 - Is 3 minutes a magic number of some kind??

All --

Having the a problem with calls cutting off after 3 minutes -- not 2:30 or
3:12, but *exactly* 180 seconds.  To illustrate, I have the following
extension:

Exten => 8005551212,1,MusicOnHold

If this extension is dialed from a Cisco 7960 (SIP), the call plays the
music, and then disconnects after 3 minutes/180 seconds exactly.

If this extension is dialed from a PSTN Zap interface, the call plays the
music, and disconnects after 3 minutes/180 seconds exactly, giving the
caller the congestion tone.

If this extension is dialed from a Pingtel Xpressa or ATA186, the calls 
stay connected after 3 min and indefinitely just fine.

Wondering what sort of value could be affecting both Zap and SIP call 
progress/timeouts, etc.  Any input appreciated.

Regards,
Dave

====================================================================David C.
Troy   [dave@toad.net]                   410-384-2500 Sales
ToadNet - Want to go fast?                        410-544-1329 FAX
570 Ritchie Highway, Severna Park, MD 21146-2925  www.toad.net

On Sun, 2003-09-07 at 16:07, David C. Troy wrote:
> All --
> 
> Having the a problem with calls cutting off after 3 minutes -- not 2:30 or
> 3:12, but *exactly* 180 seconds.  To illustrate, I have the following
> extension:
> 
> Exten => 8005551212,1,MusicOnHold
> 
> If this extension is dialed from a Cisco 7960 (SIP), the call plays the
> music, and then disconnects after 3 minutes/180 seconds exactly.
> 
> If this extension is dialed from a PSTN Zap interface, the call plays the
> music, and disconnects after 3 minutes/180 seconds exactly, giving the
> caller the congestion tone.
> 
> If this extension is dialed from a Pingtel Xpressa or ATA186, the calls 
> stay connected after 3 min and indefinitely just fine.
> 
> Wondering what sort of value could be affecting both Zap and SIP call 
> progress/timeouts, etc.  Any input appreciated.
The case of the SIP to SIP calls get routed outside of asterisk after
initial setup and therefore do not get limited. Your Zap to SIP, Sip to
Zap, and Sip or Zap to asterisk would be limited by what asterisk does.
Look for an Absolutetimeout in your dialplan.
-- 
Steven Critchfield <critch@basesys.com>

>Hi there,
>
>Should asterisk have many, many UDP ports open?
>
>[root@router asterisk]# netstat -nap|grep asterisk|less
>udp        0      0 0.0.0.0:17920           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:12544           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:10240           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:12800           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:17921           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:12545           0.0.0.0:*      19773/asterisk
>udp        0      0 0.0.0.0:10241           0.0.0.0:*      19773/asterisk
>
>...
>
>[root@router asterisk]# netstat -nap|grep asterisk|wc -l
>     447
>
>I know that the other day I had to recycle asterisk because I was getting
>"too many open files" errors in the log file (and asterisk was
dead).
>
>...deon
This may or may not be related to the SIP bug which has been open for 
some time.  You may find more of a description at 
http://bugs.digium.com/bug_view_page.php?bug_id=0000055

Or, you may have found something different...

JT