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When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

...igium.com/mailman/listinfo/asterisk-users >> > > Bill > > > William M. Conlon, P.E., Ph.D. > To the Point > 345 California Avenue Suite 2 > Palo Alto, CA 94306 > vox: 650.327.2175 (direct) > fax: 650.329.8335 > mobile: 650.906.9929 > e-mail: mailto:bill@tothept.com > web: http://www.tothept.com > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/aste...

Hello, I have started having a strange problem. Asterisk is connected via 4 analog lines to PSTN and we have SIP phones internally. All was working fine but recently each time a user calls from PSTN and when he is leaving a voicemail for someone, the caller gets disconnected after 30 secs. We have AMP installed. This is reproducible and is happening always. It seems that Asterisk is disconnecting

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated

Hi listers, I am looking for people who have used Power over Ethernet switches, primarily in conjunction with Polycom IP 501's. I've been looking at the Linksys SRW224P, since I've had good luck with the SRW224 in our office. However, Nortel, Cisco, Adtran, etc. all have an offering, all of which vary in price. I would appreciate any input people have to offer. Thanks, James

From what I know this log show everything working Perfect. Then it goes to the Welcome screen then after a long time of processing, it errors out with a 0x10000 error Any Ideas? 1005195711|so |4|00|---------- Initial log entry ---------- 1005195711|so |4|00|+++ Note that bootrom log times are in GMT +++ 1005195711|hw |4|00|Initial log entry. 1005195711|wdog |4|00|Initial log entry

I've got two Polycom 501s in an Asterisk 1.2.6/zaptel 1.2.5 with Digium connected to POTS. My legacy key system is still running in parallel. Today my 501 didn't ring on an inbound call, nor could I call out. sip show peers showed no ip address for my 501. Rebooted the phone and it's ok now. There wasn't anything enlightening in /var/log/asterisk/messages. Has anyone

I was just getting to work on fax for my * system, so I thought I would bring everything up to date since there would be some new compilations involved. yum update gave me kernel-2.6.16-1.2069_FC4 but after recompiling zaptel, I kept getting "FATAL module zaptel not found" Chased this for an hour with multiple recompiles and reboots. Finally dropped back to 2.6.15-1.1833_FC4,