similar to: Please Help - 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.

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:

Hi all, Have some one advice me on the following needs? 1) Clients need to use 50 Hard IP Phone, and 20 Soft-Phone. 2) Clients have 20 Hard IP phone and 10 Soft-phone within Operational Department. 3) The Soft-phone mostly is division head, and they wish to monitor the online status of the existing hard-phone and soft-phone through the soft-phone. The status like

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

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

Apologies if this is a resend, but I've sent this 12 hours ago and still can't see it on the list. Hi, I've just started to setup my phones with Buddy watch. Basically, it all works fine when using the simple example on the wiki: exten => 123,hint, SIP/some_sip_reg exten => 123,1,SIP/some_sip_reg BUT, what I need to do is dynamically decide where the hint checks for buddy

I need a pointer to what Fine Manual I should go read to set up my netbios name resolution, because all my attempts to use netbios names are being resolved against dns ... I have one subnet (192.168.2/24), with several Fedora 11 machines and several Windows XP machines, and a couple SMB/CIFS NAS units which also share printers. No domain, it's all just a workgroup. The WinXP machines

LogBuddy is your friendly little log buddy at your side, helping you dev, debug, and test. It plays well with Rails and plain old Ruby projects. To use it, sudo gem install log_buddy, then require ''log_buddy'' and call LogBuddy.init. It will add two methods to object instance and class level: "d" and "logger". You probably only want to use these in non-prod

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 --------------

I've just set up my dial plan to use hints for each of my extensions, and I've set up a buddy watch list on my Polycom IP-501's, however, I keep getting the following messages every 5 or so minutes: -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.1.106 -- Incoming call: Got SIP response 500 "Internal Server Error" back from

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

> 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

Is this issue in the CentOS Bugzilla or Upstream Bugzilla? I use a fully updated CentOS 5.2 box. Frequently, when I close Evolution, I get a message from Bug Buddy, the Bug Reporting Tool, that Calendar has crashed. I do not use the Calendar Application. I use Evolution for email and contacts. Then, when I try to send the information, I get an error message: Bug Buddy has encountered an error

I could have sworn I saw a post about this recently but I can't find it so apologies if this is a dupe, but is there anyway to control the order in the Polycom Buddies list? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070206/b2d3c1fc/attachment.htm

Folks, First of all, this email is sent to -users and -biz, but please follow- up to the -biz list only. I have set the reply-to, but I fear mailman will strip it off ... Please don't flame me for posting to -users, I'm just not sure who lives on -biz as the (signal/noise | net.kook | troll) factor has been pretty bad on that list lately (poor Rehan!!) ;-) Telephony Depot

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

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

There seems to be some hostility to the idea of this being a GNOME or Evolution problem: Bug 542280 ? Refuses to report bugs because gnome 2.16.x is too old View Bug Activity Product: bug-buddy Component: general Version: 2.16.x Status: RESOLVED Resolution: WONTFIX Opened by Mark Hull-Richter (reporter, points: 3) 2008-07-09 23:40 UTC [reply] On CentOS 5.2 (and 5.1 and 5.0, and Red Hat

Thanks, Henry. It is very helpful for me. I also deleted the DIAL option "r" in our dial out trunk which fixed the problem. Dial command option r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Without this option, Asterisk will generate ring tones automatically where it is appropriate to do so; however, "r" will