similar to: polycom soundpint ip650 question

Hello all, I have some polycom 430's which I'm trying to get to work with asterisk. I have them working for the most port other than one little issue. The 430's have two line appearances. I'm trying to get the second line registered to a different extension but for some reason it's not allowing me to do this. The first line will register fine but the second line never

I have a Polycom Soundpoint IP335. There are no inbound routes set to the phones yet. However, the phones are getting phantom rings. What is the legitimacy of these calls? Is there something I need to block to stop it? I believe its people trying to hack the phones/phone system but I cannot find where I read that before. Thanks, --E -------------- next part

I got em. You want em? Anyone know how I can get these to the site listed on the Wiki? Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/dfee13b3/attachment.htm

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the

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

Does anyone have an actual delivery date on the new Polycom HD IP650's? I'm getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061018/9937184f/attachment.htm

I set up hints and presence monitoring on some Polycom phones connected to an asterisk server with the expectation that the phones that are "watching" other extensions would be notified when the other extension sis ringing, in addition to the other statuses (on the phone, statuses set by the user on the phone, not registered, etc). I can see when the line is in use, and when it is

I am trying to get something working that is just not doing quite what I want. It may not be possible, but I figured it was worth asking about. The details: Asterisk 11.6.0 Polycom SoundPoint IP650 phones running 4.03 firmware. We have a queue with 4 phones in it. ringinuse is set to yes and the stategy is ringall. In sip.conf, we have notifyringing set to yes as well. Asterisk is sending

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, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 650 Data: ? P/N: 3150-11530-212 ? SD Sound ? FW: 2.1.2.0078 ? Assembly:

Anyone seen anything on the IP670 & the Color Expansion? -Matt

When you perform an attended transfer, the extension of the person transferring is displayed to the co-worker. Can I override the caller ID to display the caller's callerID during a blind transfer? Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I''m fairly new to script.aculo.us, so hopefully this is the right place to ask this question. I have the following situation (pseudo-code): <div> <Droppable 1> <Droppable 2> </div> <div (scrollable)> <Draggable 1> <Draggable 2> ... </div> Basically, my Draggables are inside a scrollable <div> element (the interface

To the volunteers of R-Help. Hello, I am currently stuck on an RStudio assignment. The assignment involves creating a double bar graph with the provided info http://math.fullerton.edu/mori/data/introstats/pennstate3.txt My professor has only gone over the very basics of RStudio and we only learned how to make a simple bar graph and labeling x and y axis. The specific directions from my

I have two scrollable div''s (overflow:auto), one with a list of elements (the source) and the other is the drop target (dest). I''ve enabled ghosting so that the drag element gets out of the scrollable box (good). Interesting, at least on Firefox, the ghosted drag ends up going ''under'' the destination div when I drag it. No amount of z-order fidding seems to

I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris

With asterisk 1.8.7.0 has been running ok for months. Now, this morning, it's crashing. I can restart it, but it crashes after 10+ minutes. It dies like this -- Executing [s at macro-stdexten:2] Dial("SIP/teliax-00000019", "SIP/176,18,rtT") in new stack == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 == Using SIP RTP TOS bits 184 == Using SIP RTP

Hi all, I''m a software developer from New Zealand. I''ve built a couple of rails sites and have some ajax experience. I''d like to develop on a rails site with the aim of making a business - is anyone else looking to do the same? The two rails sites I built recently are foopad.com and friendr.com. Regards, Ben

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave

Greetings, I'm using asterisk to connect our three locations together with a sort of inter-company auto attendant connected like this: PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk It works like this: Person picks up their phone and dials a number to get to the auto attendant (I don't have any FXO ports available on our PBX to do it the