similar to: multiple registrations with Polycom IP600

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

I'm stumped on this one and any help would be greatly appreciated. I'm just trying to get my Polycom 601 to have multiple extensions on it. For example, on line 1 I want extension 21, on line 2 I want extension 22, and on line 3 I want extension 23. Ideally I'd actually have each extension appear on 2 lines and therefore filling up all 6. I should be able to do that with the

I'm trying to figure out how to allow an extension to register more than once. For instance, I have all of these 4 line IP phones that I use with Asterisk and I would like to have a persons extension (say 101) ring at all four lines so that if the person is on the phone they can take another call, but it appears as though if you try to register the same extension more than once then the most

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Hi all - This isn't really directly Asterisk related, but has anyone successfully set up a Polycom phone to register two lines on two different Asterisk boxes? I can get the first line to register, but the second one does not. I can still place calls from that second line, which indicates to me the server, user, and secret are correct. I'm running the newest 2.6 series firmware with the

hi there I'm setting up asterisk@home and I'm using Polycom IP 500 phones. When I call a number that has a digital receptionist (i.e. "dial 1 or such and such, dial 2 for this and that...") the Polycom doesn't seem to send the extra digits. When I try it with X-Lite things appear to work fine, so I think the problem is with the Polycom configuration. Here's some

For the last few months we have intermittently been experiencing some very strange registration problems with certain polycom phones. Here is some background information: I have about 150 Polycom Soundpoint IP 600s, 601s, and 650s spread between 8 servers at different locations. Each phone is on the same network (and subnet) as the server it connects to. There is no NAT or anything else strange

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

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.

I'm having trouble with both my Polycom IP600 and IP500 disconnecting calls to the PSTN after about 1 hour. The below log is of a phone call that lasted 1hr 39mins which is my record so far. I cannot figure out what is causing the call to terminate and I am hoping somone on this list can help me. In this example both the phone and the asterisk server have public IP addresses so NAT shoul not

Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never

Hello, scenario: OK 1. created vacation script (see below) OK 2. send message from user at anotherdomain.tld to user at domain.tld => got auto vacation response (2010-02-11T19:42:13) OK 3. send message from user at anotherdomain.tld to user at domain.tld => no response (is ok because ":days 1" settings) (2010-02-11T19:47:46) BAD 4. send message from anotheruser at

There is a samba compiled without winbind support, with the following options configured: workgroup = MYDOMAIN security = domain allow trusted domains = yes add user script = /usr/sbin/pw useradd %u -m -Y -M 755 When a Windows user MYDOMAIN\john connects to the samba server, he is mapped to the Unix user john. If there is no Unix user "john", it is created by the add user script. How

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, i've setup sieve script which send notify to user (user at domain.com). But user at domain.com is alias user at domain.com and user at anotherdomain.com. LMTP delivers notification only to user at domain.com but not to user at anotherdomain.com. How to force send notification over smtp (which will be delivered to both) or i must use to=user at anotherdomain.com in notify ? Thank you

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

Did you try to boot without lan just the power ... I've had this same problem to and rebooted the device without lan connection -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Louis-David Mitterrand Sent: woensdag 26 januari 2005 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The

Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio