similar to: Asterisk 1.8 redial polycom ip600

Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? Thanks, -- Lord, protect me from your followers.

This Polycom phone seems to be one of the best on the market for sound quality and features. I have seen on the list that some people have gotten the IP 600 to work with Asterisk. Does anyone have the details of how to get this working i.e. XML phone config files, and any thing else I might need to know. Thank You, Chad Cowan -------------- next part -------------- An HTML attachment

Hi, I'm testing a Polycom IP600. With firmware version 1.1 the phone reboots at any time. With firmware version 1.2, the first reboot was an endless reboot. Then I moved the phone to another lan port, then it worked fine. Then I installed again in the initial lan port and the phone works well. However after some time of inactivity (1 hour?), the IP600 stops to send and receive calls.

Hello list: I have a very odd problem. Seemingly randomly, my Polycom IP600 phones will ring without a call being placed to it. That is to say, a random phone will ring. Nothing shows up under Caller ID. Even the buttons that light up to show an incoming call do not light up. If you pick up the handset, you can hear the phone ring through the speaker. Hanging up the phone makes it stop

Hello, I have configured my polycom ip600 and ip500. The phone works well. But the clock is wrong and flashes the whole time. Drives me nuts! I have set the time offset on the DHCP / boot server. 36000 (I'm in Australia!) It doesn't seem to help... Mike

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Hi all, I dont know why, but if I plug my PC inside the 'PC' slot on my polycom, this is not working. (Polycom IP600 is online on the net.) I'm using normal network cables. (I see jumpers behind the phone... do I need to play arround with that?) Any help would be appreciated. -- Alexandre Leclerc

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks.

Hi All, Can someone please advise me about configuring my Polycom IP600? I have an account with a SIP based IP Centrex provider. The basic SIP info and line 1 config points to them. That's working fine. I'd like to register line 2 with my own asterisk server. I've tried putting the basic registration info in the line 2 configs...but the phone never registers. Not certain how to

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a

Does anyone have the Admin password for the phone in order to change configuration Roman

here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits

Hello, I would like to add area code to local numbers, it worked like a charm on Asterisk 1.8 but does not work on Asterisk 13.11. Extensions.conf; worked before on Asterisk 1.8 ; Adding Area code to local numbers exten => _9XXXXXXX,n,Set(CALLERID(all)="$CallerID" <3818008000>) exten => _9XXXXXXX,n,Dial(SIP/voip-truck/1381${EXTEN:1},80) Any ideas? Thanks, Motty

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

On 10/28/2015 06:37 PM, Pete Mundy wrote: > Hi Motty, > > Isn't the whole point of the nonce in a SIP registration to ensure the > secret doesn't go on the wire in plain-text? Is this not enough, or > are you looking to hide the username too? > > (if so, fair 'nuf, just wondering why :) > > Pete > > Ps, if so then I think TLS is the missing part of