similar to: Polycom registration errors

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

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

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

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

Hello, In our SIP network, Asterisk is the central PBX, and it routes calls to the PSTN thru a Cisco Router - IOS 12.2(11)T9. If a client softphone calls directly via Cisco to the PSTN, the call works successfully. If the client softphone calls via Asterisk to other SIP internal extension, it work fine too. The problem is when a client calls an Asterisk extension, and Asterisk transfers

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

I have a similar setup to you and get the same message regularly. I don't think it's the cause of your problem. I did some research on it a while ago: IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk (correctly) uses codec 19. The router can be configured to use 19 also, but I didn't bother. I'm sure somebody will correct me if I'm wrong about

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

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

Hi I want to mount the a share of a windows 2003 Domain Controler, but I'm not been able... Does any body knew a solution for that problem? Thanks.... here is the error message: <snip> root@r-pad:/mnt/max-ecom# mount -t smbfs -o username=mnicolas,password=sucker //192.168.200.99/Multimedia /mnt/ cli_negprot: SMB signing is mandatory and we have disabled it. 11688: protocol

Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit confusing with SIP and Line tabs. I have put this on the web interface: SIP > Outbound

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.

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