similar to: IP601 Expansion Module HELP!!!

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

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

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,

Hi Polycom experts, I'm having a problem getting changes to the Polycom IP 601's (mac)-directory.xml file to update the button list on the phone. If the phone is newly provisioned (i.e. if I "Format File System" on the phone) then the new list will show up on the buttons, but of course this is pretty drastic way to do it. - Environment: Asterisk test setup with 7 phones,

Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

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 List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a "command" which im guessing is the program code like the rest of the Avaya systems works. Has anyone

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

Hi list, Ive got a server with 6 ports on it (4+2 port card) we have a DS3 delivering all voice DS1's to us. Carrier has a trunkgroup for the first 8 span (we only have the first 6 plugged in right now). Everything works fine until we fail the primary D channel (D's are on 24,48) the secondary then picks up and outbound calls do not work, if we reboot Asterisk the D on 48 comes up and it

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

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

my view contains a call to a partial: <%= render(:partial => ''item_list'', :collection => @keyword.synonyms, :locals => { :action_delete => "removesynonym", and_some_other_stuff }) %> _item_list.rhtml contains: <%= link_to ( image_tag(''/images/deletebutton.png''), { :action => action_delete, :id =>

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.

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:

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

Hello guys, I''m a scriptaculous newbie (I started working with it only yesterday) and I have already the first problem. I''m trying to implement a drag & drop list (fallowing the shopping cart example http://demo.script.aculo.us/shop) and I''m almost done but after dropping an item on the target div I need to refresh the page to see that the item has been moved.

Hi Jay - I see the slow, small accesses using an older clang [Apple LLVM version 7.0.0 (clang-700.1.76)], but this looks fixed on trunk. I made a change that comes into play if you don't specify a particular CPU: http://llvm.org/viewvc/llvm-project?view=revision&revision=245950 $ ./clang -O1 -mavx copy.c -S -o - ... movslq %edi, %rax movq _spr_dynamic at GOTPCREL(%rip),