similar to: Polycom IP 501 buddy list: Got SIP response 500 "Internal Server Error"

Hi, I found your post on http://threebit.net/mail-archive/asterisk-users/msg04580.html I am having the exact same issue with the Polycom IP601 (SIP version 1.6.6.0036) with Asterisk 1.2.7.1. I was wondering if you found any solution to it. I would really appreciate if you could share your solution. Thanks, Khairul. BELOW, THIS WAS YOUR POST 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, 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:

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

I've been noticing that my group of Polycom IP 501 phones seems to randomly reset themselves nearly every night (I guess it usually happens at night, since I've never seen it happen while I've been at work during the day).. When I say "reset", I mean, the hands free volume and ring volume are set to the default and the call logs (received calls, missed calls, placed calls)

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.

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

Hello, I have so far Polycom 501 and 403 which displays the label of a key and the name of a buddy near its key. Today I received new 330's and they do not display the name, only "1" & "2". Besides that they work correctly. Anyone has an idea, or is it a known "feature"? Thanks! __Yehavi: -------------- next part --------------

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

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

Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".....I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add <bw>1</bw> manually to the speed dial file it doesn't change anything. The buttons work well for

JR Richardson Engineering for the Masses> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of asterisk-users- > request at lists.digium.com > Sent: Saturday, March 01, 2008 12:00 PM > To: asterisk-users at lists.digium.com > Subject: asterisk-users Digest, Vol 44, Issue 1 > >

I am having a couple of (unrelated) problems with my polycom 501. 1. The buddy watch is just not working. It tells me that everyone is online, whether or not they are. Here is an example directory entry for one of the peers (whose phone is not registered). <item> <ln>F</ln> <fn>J</fn>

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

Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The

> I ran into this problem with an early batch of IP650s. Polycom's firmware > version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth

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

Hi list, There are many Polycom experts on this list -- hopefully someone has a solution. With *several* versions of Asterisk 1.4.x, doing a reload of Asterisk causes the Polycom 601 phones to start dumping these messages to the CLI. -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.x They continue on until we force the devices to reboot from the

Hi, folks: I just took delivery of new PoE cables (PN 2200-11077-002) for our Polycom phones here in the office. I have the following to report, for the erudition of existing Polycom users and those considering purchasing Polycom sets from the set of models noted in the subject: 1. The cables do indeed work (hooray). 2. The cable has a module on it about a third of the way from a female end.

Hi all, I've been messing around with the g729 codec in some phones I use and had made all phones use the codec for all calls for testing purposes. The problem is when I attempt to dial out on my Polycom IP 500 (test happens to be calling my cell phone) I can only hear sound coming one way, I recieve nothing from one user, just silence, yet I can talk one way perfectly. Now I tried the same

I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone