Re: [asterisk-users] oddity with TDM400P / Asterisk setup

2006-08-31 Thread Ted Wallingford
Hey, in case anybody has this problem.  I went out to the location of the Asterisk box and found two problems. The default gateway, a sonicwall, was sharing an ip address with a wifi point (doh) and a patch cable plugged into his switch on both ends (doh again).   --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Aug 30, 2006, at 5:39 PM, Ted Wallingford wrote:Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports.  There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything.  The * version running is 1.2.7.1.   All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort.The problem, which began this morning, is very elusive.  Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call.   At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell.  Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang.Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem.  I also attempted to move the server to another port on the switch. No improvement.  Anybody have a problem like this?--Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com  --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Jul 13, 2006, at 3:02 PM, Warren (mailing lists) wrote:Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work,etc. off of this list? This is not the place to discuss yourexperiences with _any_ company, it's a place to talk about Asteriskand using Asterisk.Please move flamewars and similar discussions to some other forum. I agree with you!Which place is in your opinion the right place?As long there is no other place, such messages will always pop up. How about the Asterisk-biz list?W___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


[asterisk-users] oddity with TDM400P / Asterisk setup

2006-08-30 Thread Ted Wallingford
Hi List,I am working with an Asterisk server running on Fedora Core 4. It has two TDM400P cards installed. There are 6 trunk ports and 2 (unused) analog line ports.  There are 5 Polycom SoundPoint 501 SIP phones connected to the server, and a Linksys 24-port powered switch connecting everything.  The * version running is 1.2.7.1.   All of the ports on the switch with voice devices, including the server, have a service class of 5, while non-voice devices are connected to other ports that have a service class of best effort.The problem, which began this morning, is very elusive.  Calls-in-progress from zap-to-sip or sip-to-zap or sip-to-Asterisk will drop at odd times during the call, anywhere from 2 minutes to 15 minutes into the call.   At the same time the call drops, my SSH session to the server will hang. After 10 to 15 seconds, the output and input from ssh session appears on my terminal and I am able to resume working in the shell.  Zap-to-Asterisk doens't seem to cause the problem. Only when I dial through to a SIP device does it seem to hang.Top reveals nothing out the ordinary, utilization wise, the disk has plenty of free space, and the arp cache doesn't ever indicate a duplicate IP address with the server's NIC, which I thought might have been the problem.  I also attempted to move the server to another port on the switch. No improvement.  Anybody have a problem like this?--Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com  --Ted WallingfordBest Technology Strategy LLC440-864-6084 phone440-815-2083 fax[EMAIL PROTECTED]http://www.btstrategy.com On Jul 13, 2006, at 3:02 PM, Warren (mailing lists) wrote:Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work,etc. off of this list? This is not the place to discuss yourexperiences with _any_ company, it's a place to talk about Asteriskand using Asterisk.Please move flamewars and similar discussions to some other forum. I agree with you!Which place is in your opinion the right place?As long there is no other place, such messages will always pop up. How about the Asterisk-biz list?W___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users