similar to: Issue w/ Polycom 501 phones in a queue...

Howdy - I've been noticing a problem where I only receive a single call, before other calls go to voicemail. This only happens when the user is on the phone. I have the polycom 501's setup for 2 lines per key and 2 line keys for the first registration, which should allow for multiple calls. Anybody have any ideas what is going on? thanks Rolf Brusletto

All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or

I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone

We're looking to purchase new phones for Asterisk. There are a limited number of new Polycom 501's on the market, mostly refurbished available. Can you recommend a replacement phone? What ever model replaced the 501? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

Has anyone else encountered this? I have a fairly large installation (~50 phones, almost all Polycom 501's and a handful of 601's. We're running into a number of phones on which the outbound voice (Polycom phone user doesn't hear any problems, but the other end does) is breaking up occasionally -- enough to be noticeable and make you say "what?". In each case, rebooting

Hi List, Not sure if this is the place for this so here goes ... We have a number of Polycom 501's connected to our * box and they work great. Some of our users have added a few entries into the directory on the phone. The problem is on those particular phones they now sometimes get "resource full" on the phone when accessing the directory. No central directory was configured.

Hi, everybody: I have looked at the Polycom entries on www.voip-info.org, and they're outdated and convoluted and full of errors. All I want to do is get my Polycom 501 to register with a working Asterisk server. I want to do the configuration locally on the phone through the console. (The server works with an Xten X-lite softphone.) Has anyone done this? What do I need to do? Thanks,

Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom

Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Thomas Johnson > Sent: Friday, January 20, 2006

Hi, Does a parameter exist for a queue to delay ringing/sending a caller to all agent phones after the previous call is answered by an agent? My queue ring strategy is set to ringall. I am using Polycom KIRK wireless DECT SIP phones. And it looks like the KIRK wireless server may need a split send to realize all wireless phones are no longer ringing (busy) after 1 call rings & is unanswered,

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

Hello- We've got a patched-together extensions.conf that's barely working for us, and we need to get real about using Asterisk. We've got a couple of remote workers with Polycom IP-601 phones, and a single asterisk server, using a couple of incoming DIDs from teliax and sixtel. Does anyone have a good extensions.conf that they'd be willing to share, that provides a

I'm having a problem with polycom ip601. If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom rings, and stops ringing as soon as I hang up. But if the phone is called via a queue, the polycom continues to ring long after I've hung up. Other phones in the queue (grandstream, cisco) don't have this problem, and stop ringing properly when I hang up. polycom bug

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Hi, folks: I just took delivery of new PoE cables (PN 2200-11077-002) for our Polycom phones here in the office. I have the following to report, for the erudition of existing Polycom users and those considering purchasing Polycom sets from the set of models noted in the subject: 1. The cables do indeed work (hooray). 2. The cable has a module on it about a third of the way from a female end.

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

Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia

All - I've a new system, that since it's been in production, has seen a few issues, that look like they should be fixed by upgrading asterisk @ home to the latest version. I was curious if anybody out there can tell me their experiences with this, and what to expect. Thanks, Rolf Brusletto

Hi all, I'm having a problem with a polycom 301 not picking up a ZAP call. Below is the CLI output of the call. I have: TDM400 with 2 FXO lines Asterisk 1.2.14 Polycom 301 When I dial the first ZAP line, I choose an extension that rings the polycom, polycom rings and I can pick it up and the call is bridged. When I call my second zap line, the polycom rings, but I cannot pickup the

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power