similar to: PoE switch recommendations?

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

We are currently considering the Linksys POE switch for a small Asterisk office deployment. There will be no separate wiring closet to put it in. Can anybody tell me if this switch has a loud fan? Users would not be able to tolerate a loud noise close by. Otherwise we will go with a fanless desktop switch. Thanks, -- Andres

I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The Linksys SRW224P behind me right now would be objectionable, I'm sure. -------------- next part -------------- An HTML attachment was scrubbed... URL:

2011/11/30 Mike <list at net-wall.com> > Hi Olivier,**** > > ** ** > > It if occurs only on the sidecar, I would imagine this is either a > defective sidecar/Polycom phone, or a defective PoE switch not giving > enough power. Changing PoE port would eliminate of confirm the PoE port > being the issue, but I?m betting on a Polycom defect.**** > > ** ** > >

> > Allen Casteran wrote: > > Anyone know what the POE draw is for the Aastra 480i phones? > > We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. > > A Cisco 3560 switch will do 15.6 watts on all 24 ports. > > Just trying to find out if we need that much power. > Drew wrote: > According to Aastra tech support, 5 watts

Hi, I've looked high and low for any changes that streaming MoH needs on Asterisk 1.4 (.21), followed NerdVittle's article about it (http://nerdvittles.com/index.php?p=92) yet nothing worked. After creating dir stream/ and touch stream.mp3, here's my musiconhold.conf [stream] mode=mp3 directory=/var/lib/asterisk/mohmp3/stream stream =>

Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me

Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR00003.wav FR00003.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is

Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments?

Spinning off from another topic...what are people using for QoS / Shaping? I'm using Wondershaper script with OK results...but I'd like better. Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091008/96b14a3e/attachment.htm

Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

Hi List, Does anyone have experiences to relate on the various Asterisk-based PBX appliances out there? Like the Aastra 160, Digium S844i, etc. Do the Epygi Quadro and Grandstream GXE also use Asterisk? Thanks, Femi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm

Hi, according to discussion on asterisk IRC, where people said, that macros will be depracated, I tried to migrate from macros to contexts and Gosub but if I try to use gosub in extensions.ael, ael compiler complains, that I shouln't use Gosub app, but I can't find ael keyword, that will be Gosub equivalent, or can I ignore this ael warnings? thanks PJ LOG: lev:3 file:pval.c

How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Thanx, Daniel Arohuanca Lagos +51 1 3594122 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080722/f9612f97/attachment.htm

Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register =>

Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. -------------- next part -------------- An HTML attachment was scrubbed... URL: