similar to: Polycom-Asterisk hints/presence

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

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

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

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! A few months ago I needed some help for the following issue: .) a call comes in .) Person A takes the call and does an attended transfer to Person B .) Person A hangs up the phone without waiting for Person B taking the call .) the caller get lost at this point !! At this point the attended transfer should go into a blind transfer. The phone of Person B should still be ringing and the

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 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

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at

Greetings, The Polycom SIP 1.5 Admin Guide says this: "3.1.8 Connected Party Identification Where possible, the identity of the remote party to which the user has connected is displayed and logged. The connected party identity is derived from the network signaling. In some cases the remote party will be different from the called party identity due to network call diversion."

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

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

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

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

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

Has anybody got the polycom acd function to work? I have the following setup: Debian 3.1 - 2.6.8 linux zlib-1.1.4 libpri-1.2.3 zaptel- 1.2.6 Asterisk - the bweschke/polycom_acd_funtions branch version - I get one error when doing a make install about needing a newer version of libpri and zaptel, I got the above versions from asterisk.org, are there newer version anywhere else? In the sip.conf

Hi, Is there ANY way for me to see the status of the Polycom DND buttons in the Asterisk hints? I`m using the BLF buttons to see the status of other people`s lines, and DND should logically be somehow reflected (I don`t care as much about Polycom showing the BLF button as DND, but I do care about Asterisk hints showing it in the CLI). Right now, a Polycom phone on DND shows up as being

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

Anyone know how to override the 99 message limit in voicemail? (yeah, we have a public VM that gets that many a day). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050812/1bff12e4/attachment.htm

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