similar to: polycom queue bug

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,

How do you disable call waiting on Polycom IP601 phones? I've looked through the user and admin guides and can't see anything about disabling it. -Dan

I was reading a Polycom brochure and it appears that there is really no special receptionist console and the console is basically a IP601. Is this correct? The only difference is to purchase an expansion module in order to have more shortcut keys for the girls. So, apart from the hardware, as far as the dialplan is concerned, do I just treat the receptionist console as any other extension? Are

Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. <dialplan dialplan.impossibleMatchHandling="2"> </dialplan> (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size

Make sure you set nat=yes for the sip user. Asterisk will then send replies back to the source IP address, rather than what's in the Via: header. > -----Original Message----- > From: Warren (mailing lists) [mailto:warren-lists@icruise.com] > Sent: Friday, October 27, 2006 5:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Taking a

What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

Hello people, I am having difficulties figuring out the POE power draw in watts from a Polycom IP601. I want to know how many IP601's can be powered from the Cisco Catalyst 3560G-48PS. The IP601 wallwart has: Input 120VAC 60hz 19W, Output 24VDC 500mA. I assume the output is appropriate value to figure out how many phones can be powered. The Cisco 3560 datasheet says "the 48-port PoE

I am working on Polycom IP601 console with expansion module. I want to put on the BLF (busy lamp field) feature on all the contact/speed dial names I put on the console but I could not get it to work. *CLI> core show version Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on 2007-11-20 05:26:15 UTC *CLI> sip show subscriptions Peer User Call ID

Does anybody happen to know the input power specs for the Polycom IP 500 and IP 600? We've mixed up our power supplies and we've got a whole box of them and can't figure out which go to the Polycoms. I would rather not kill the phones by trying random ones....

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite the problem getting the Polycom SP 501 to send *any*

Hi, I have an Asterisk 1.8.2.3 installed (public IP) with a peer (Polycom IP601) installed behind NAT. When the peer makes a call, it's working without any problem. But when a call is coming back, it ends up with a Got SIP response 400 "Bad Request" back from xx.xx.xx.xx where the xx.xx.xx.xx is the public IP of the peer. And the call drops to the voicemail (congestion at peer

Hi, I found your post on http://threebit.net/mail-archive/asterisk-users/msg04580.html I am having the exact same issue with the Polycom IP601 (SIP version 1.6.6.0036) with Asterisk 1.2.7.1. I was wondering if you found any solution to it. I would really appreciate if you could share your solution. Thanks, Khairul. BELOW, THIS WAS YOUR POST Polycom IP 601 Buddy Watch problems

One more Polycom IP601 question please (sorry for the long intro here to document) ... In order to closely approximate the behavior of the previous telephone system that many of the users are familiar with, I have set up call parking like this: - features.conf [general] section contains: parkext => ** ; What extension to dial to park parkpos => 10-11 ; What

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

This is a SER/Polycom question, but I hoped we may have some SER guru's here... I have a series of Polycom phones that are tying to register with OpenSER. The phone sends a REGISTER message, and OpenSER replies with Unauthorised (all normal). The phone re-sends the REGISTER with the credentials, and OpenSER sends Ok. Here's where it goes downhill. The polycom's appearance display

I have PolyCom phones in one office working perfectly, but in another office with a new subnet, new server, new everything, the time does not work. Everything else about the phones seems fine, but the time. If you look at the internal webpage in the phone, it shows "clock". Our server, which is configured to allow others on the net to get their time from it, and it in turn gets its

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device

Hey everyone, I haven't seen an issue quite like mine, so I am hoping anyone who used the Polycom 601's may have an idea. We are going to be switching our office over to Asterisk. All the phones are going to be 601's, I am going to set up a boot server, but for now I am just going to test everything on one phone. My question is I have the phone registered in Asterisk (phone icon