similar to: Pushing info to a Polycom phone - from outside of the local network

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.

I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my

We are having an issue that I have been unable to figure out how to resolve. I think its related to the Polycom Phones and not the Asterisk configuration, but I'm not positive. We have several Polycom 500/501/601's on both a LAN and at employee homes. The problem we are having is if our internet connection goes down the Local LAN phones loose their connection to the Asterisk Server.

Hi, I have been trying a (really simple) push application for the Polycom microbrowser, using a Polycom 650 with 3.2 firmware. I can't do anything, I always get "Push message cannot be displayed" back from the Polycom phone, and all I am sending is the Polycom example : <PolycomIPPhone> <Data priority=?critical?> <h1> Fire Drill at 2pm </h1>

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

Hi, I've had a functioning Asterisk system (1.2.18), which I haven't reconfigured in any way, that is just now refusing to forward calls. I only have Polycom phones. When I use the phone's forward feature (forwarding the phone with extension 204 to extension 206, which used to work as recently as yesterday) I get this in the console: "called sipreg-12344". No ringing,

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

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

I have meetme working with BLF on polycom phones however when meetme is not actually being used by anyone the 'status' of meetme becomes "idle". Which the Polycom phone sees and produces a clock symbol and FLASHING red LED. Are there any 'tricks' or work-arounds to change this status to something that does not blink the phone's LED making it look busy when meetme is

Hi I am struggling with hardware choices to get started with. My options are narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G. of importance is: - functionality / integration with asterisk - headset functionality and use - voice quality - build quality Is there much of a difference between Polycom and Cisco? Scanning the group it looks like there may be slightly more

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

All, Has anyone experienced a problem with the Polycom 650 phone freezing when you try to do a transfer? I am running asterisk 1.2.20.1 with Polycom SIP version 2.1.1.0052 and boot rom version 3.2.3.0002. I have Polycom 501 phones that work perfectly with the same software versions. The 650 phone; when I hit transfer the caller is placed on hold and the phone is still operational. As

PBX: Asterisk 1.4 Phones: PSTN phone connected to TDM400 X-Ten Lite Polycom 430 Scenario Polycom 430 = User1 User2 calls User1(Polycom 430) asks to be transfered to User3 User1 does an attended transfer using the trnsfr button on the polycom User2 is placed in music-on-hold User3s phone rings. (So far so good Right?) User3 picks up the phone to answer User2 only to

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

I have a Polycom IP600 serving as a receptionist phone. We developed a call manager via c/gtk that runs on a touchpad. It allows them to transfer calls, transfer to voicemail, page, etc. The problem is this: When paging another phone from the touchpad, I have to open a channel to the receptionist phone. This rings the receptionist phone. When she picks up, it then pages the desired person. This is

I'm looking to send calls to a phone with no callerid data whatsoever shown on the Polycom as far as missed call. The specific application for this is that I have a 50 phone install with some being used for paging. Paging works perfectly, but the problem is that for every page there is a "missed call" shown on the screen. I have access to the Polycom phone.cfg file, and

> > Jason Brown wrote: > | Anyone have experiece with polycom phones? > | > | I am experiencing a really weird problem. In an office where I have > | the following extensions: > | On the Polycom phones, when I want to dial from extension > 100 to any > | extension 120 or above, or dial out, it dials just fine. If > I want to > | dial from extension 100 to

Can someone please help me with a problem that I seem to have with this Polycom 601 phone. It will not see my TFTP server and keeps saying "Could not contact boot server, using existing configuration". I have Linksys phones that use the TFTP server without any problems but this Polycom will not see or use it. Please Help. -------------- next part -------------- An HTML attachment

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

>From their website, the key difference between the polycom 500 and 600 phones is the number of "lines" they support. What does this mean in terms of asterisk? Do I have to have a seperate extension for each of these lines or ? Also, slightly off-topic, how does the 500 POE "optional" cable work? Is this similar to have a power box on your desk, or is just a differently