similar to: SIP phones at multiple locations

Displaying 20 results from an estimated 20000 matches similar to: "SIP phones at multiple locations"

2006 Mar 05
6
Polycom 501 power over ethernet
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
2007 Jan 18
1
RE: Polycom buddies question
A follow up (late better than never) Finally had time to sit down and look at this sip.cfg <keys key.scrolling.timeout="1" key.IP_500.31.function.prim="BuddyStatus"/> This turns the Services key which I never use on a 501 into the Buddy Status. It even works while on a call. One touch! Bill ________________________________ From: Bill Gibbs
2006 Dec 07
2
Polycom buddies question
I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip
2005 Jan 04
6
Polycom Buddy Feature
Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP
2007 Jan 23
2
Asterisk 1.4 & Polycom buddy status
I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris
2007 Mar 28
3
PoE - IEEE 802.3af
Hi, I'm not clear on how to use Power--over-Ethernet, specifically with Polycom phones. What I understand, is that by buying the Polycom 501 with the 802.3af cable bundle, I simply connect my phone, through the Polycom provided "special" RJ-45 cable, into a PoE capable switch, and voil?! Is this true? And if so, what happens when the Phone doesn't connect directly to the
2009 Mar 23
2
Polycoms and BLF
I'm trying to get the BLF to work correctly on my Polycom phones. I have the buddy watch working correctly, but can't get the BLF to change based on the state... Example: When an extension is ringing, I get the same 'red light' that I get when the extension is actually in use... I was wondering if anyone had any experience with getting the Polycom phone to
2007 Apr 19
2
Polycom SIP Phones On LAN can't register without WAN (Internet) Access
We are having an issue that I have been unable to figure out how to resolve. I think its related to the Polycom Phones and not the Asterisk configuration, but I'm not positive. We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server.
2005 Feb 24
1
Bug in SUBSCRIBE handling : running out of RTP ports
While trying to deploy a bunch of Polycom IP 500 phones, I ran in to the following. I limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty soon Asterisk ran out of RTP ports. Traced the problem back to how * is handling SUBSCRIBE. A sip structure is allocated as soon as a request is received, which also allocates RTP ports. Normally, this is not a problem as the structure
2007 Jan 16
2
Polycom IP601 - some hints working, not others?
Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,
2006 Feb 22
4
Polycom IP 601 Buddy Watch problems
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco.
2006 Mar 23
1
SIP - Problem with audio clipping
Using a SIP connection with a CLEC, the downstream (received) audio is perfect when the mute button is activated on the phone. However, when there is upstream audio (i.e., talking or even breathing into the microphone), the downstream audio is cut off. It's kinda like having a half-duplex audio connection. When I divert outgoing calls to another provider, these calls are fine.
2006 Feb 24
1
Polycom IP 601 Buddy Watch doesn't work after Asterisk reload
Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:
2006 Apr 14
2
Polycom 501 resource full problems ...
Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.
2006 Oct 10
2
RE: Welcome to the "asterisk-users" mailing list
Polycom 601 with Sip 2.01 Anyone using Sip 2.01? I have upgraded my phones and now presence no longer functions. Buddy list shows all phones online but status does not change when someone is on a call. Also blf does not function. I am using trixbox, 1.67 was working fine on the same box. Any ideas?
2006 Jun 19
4
Polycom Buddies in 1.6.6
All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.
2005 Dec 18
12
ACD with polycom ip phones
Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com
2008 Feb 29
1
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
2006 Feb 24
1
Call quality problems
I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. The WAN comes in from the Cisco IAD and into a LAN switch (DLink DGS-1005D w/ 802.1p) where the two public IPs are switched to different devices. One device is a FireBox device controlling a separate LAN with VPNs. The other device is eth0
2006 Mar 06
1
Buddy watch?
Hi, I am using Polycom 501 and I came across a problem. As soon as I have incominglimit=1 in sip.conf, which is necessary for buddy watching, I cannot transfer calls. On the console it tells me: Call from user '3052' rejected due to usage limit of 1. Can someone please tell me how to get around this problem? (I don't know if this is relevant, but in the phone.cfg file, I have