similar to: Strange noise - Polycom

I'm sure this was asked before, but I can't seem to make this work... If a customer dials one of our DIDs, and the operator transfers that call to another employee, the Caller ID doesn't seem to do what I would expect it to. I would expect it to show the original caller's ID. Example: John calls in from the outside using (213-555-1234) and he calls into the asterisk system

> I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with > polycom phones as other devices receive my multicast just fine. > > Is there something special to do to get multicast working with polycom phones? > (other than enable multicast on the actual phone). Didn't see if anyone had answered you or not on this, but Polycom uses their own form of MulticastRTP. It

Hi Lately I've been having problem with the handsets on the polycom phones. The agents are complaining that the humming/buzzing can be heard on random days. When the agents touch their computer the buzzing may increase louder or reduced. Also when they grab the handset cable, the buzzing noise is less loud and is loud again when I let it go. I have tried changing the handsets, problem not

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

Hi: I am working on a new system with a TDM-400P card with three FXO modules and one FXS module. The system has been in place for a week. Users are complaining of echo problems. I have noticed this echo myself. It varies in severity. It is sometimes bad enough to make it difficult to converse, but the users find it generally unacceptable. They miss their old phones, which just worked. As you can

Anyone who has experience with the Polycom 330 know if the speakerphone is loud enough to be heard in a 20 foot x 20 foot room? The context is a classroom where announcements will need to be made. The phone will be wall mounted at the front of classroom. Thanks, Matthew Brothers

I know how to remap a key on a polycom 301 and 501.... But does anyone know of a list of mapping keys? For example, the Do Not Disturb on a 301 is #23. I got that one by just guessing though. Thanks, Rob

I've been able to page to a specific phone (intercom type of thing), but I'd like to have a macro or agi that pages all phones but first checks if their on the phone. It looked like there used to be a pageall.agi type of script on the wiki, but that link isn't valid anymore. Does anyone have that script, or something else that would work? I would just do SIP/1000&SIP/1001, but

Hi Guys, I have some Polycom 601's here. It is super annoying that the phone every so often beeps to let me know that I have a VM. Is there any way to turn that off ? (I just want the red led to blink that there is a VM). Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We've been running g.722 in asterisk 1.6.09 for awhile now,

Hi, One of my users is in trouble with his polycom phone hooked to an asterisk server. The phone works fine for a few days, and then disappears from the registered sip peers in asterisk. The user is able to place outbound phone calls, but can't receive incoming calls until the network plug is unplugged/plugged. Working line XXYYZZAA24/XXYYZZAA24 10.50.5.186 D A 5060

Welcome to our hell. We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally got Polycom to issue a hotfix firmware version. I'll be happy to share it with you offlist, just email me. Officially Polycom will fix the issue in 5.3 in a few months.. Thanks David On Mon, Mar 9, 2015 at 9:34 AM, Andrew Colin <andrew at convergedgroup.net> wrote: > Hi Guys, > >

I am using 11.17.0 - and MulticastRTP. Doesnt seem to work with polycom phones as other devices receive my multicast just fine. Is there something special to do to get multicast working with polycom phones? (other than enable multicast on the actual phone). Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I was wondering if anyone has had much experience with Realtime Asterisk. I like the ability to setup my extensions and voicemail boxes in MySQL, but I have a huge worry. What if MySQL crashes. I played with rtcachefriends, but can't seem to find a way to have asterisk store the extension information to ensure the phones will continue to work even if MySQL has a hiccup. Any

On Tue, 4 Jan 2011, Andy Graybeal wrote: >> The Polycom 321 has not been EOL'd and supports VLAN. It is, however, >> lacking a 2nd ethernet port if you were to go that route. >> >> -M >> > Thanks for the response Mark. I see the 331 has two ports and the same > features as the 321. > > I'm wondering what phone would be best being used as an

I am having a strange problem with polycom 501 and dailing. I've read the archives and no one there specifically mentions this problem, so I thought I'd ask here. The problem is that when the user picks up the receiver or pressed new call, sometimes they will enter a number (for example 95072091234) and in the middle of the number the cursor might jump back one digit. So the call above,

I'm trying to duplicate a behavior we had with our old avaya system, and I've come across Auto Answer (Ring Answer). However, its not quite the same yet. Right now, when I dial **5053, it will add the SIP header for Ring Answer and it will call 5053. The phone auto pickups just fine. However, we need that call to be muted. If you were to call into a meeting, we wouldn't want them to