JR Richardson
2008-Feb-29 15:17 UTC
[asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses
Bill Andersen
2008-Feb-29 19:13 UTC
[asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for over a year!) In my case, the Polycom 601 actually reboots when we page! When it comes back up, I have a phantom "meetme" on the Asterisk system and none of the sidecar lights are correct. Sometimes, they simply stop updating completely. Just FYI, go to the CLI and type "meetme". You'll get the conference ID and the number of users. Then, type "meetme kick <confID> 01" Using, of course, the conference ID. The "01" is the user that initiated the meetme. So, when you kick 01, the rest go away politely! This keeps us from having to restart Asterisk. We are on Bootrom 3.2.3.0002 and SIP 2.2.0.0047 as of yesterday and we STILL have the problem. Our setup is one Polycom 601 and 25 Polycom 501s that are being paged. The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. Our next plan is to get a 650 and see if it can handle the traffic. Bill> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of JR Richardson > Sent: Friday, February 29, 2008 9:17 AM > To: asterisk-users at lists.digium.com > Subject: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, > Interesting Issue > > Hi All, > > I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly > Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached > sidecars and Buddy Watch enabled monitoring all other SIP phones. > > The problem occurs when a group (all SIP peers) Page is called. Not > always but sometimes when the Page is executed, the IP 601 will become > unreachable from Asterisk. So when the receptionist hangs up the > page, the BYE doesn't get back to Asterisk to release all the Page > channels so they stay open. I have to restart Asterisk to release all > the open SIP Channels. > > What I think is happening is when all the SIP peers are paged, > Asterisk sends 60 hint notifications to the IP 601 and the phone is > overloaded and can't respond to SIP POKE or process the BYE message > back to Asterisk properly. > > I'm wondering if I upgrade to a new IP 650 with a faster processor, > will this eliminate the issue? > > Has anyone experienced this or have ideas for resolution or further > troubleshooting? > > Thanks. > > JR > -- > JR Richardson > Engineering for the Masses > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users