similar to: Phones that work well through NAT

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

I have upgraded the GXP-2000 to the newest firmware 1.0.1.12<http://1.0.1.12>and the phone is much more usable However, I still have two slight sound quality issues: 1) There is static on the line at all times. It is not that noticable to me, but when I make calls out the PSTN the person on the other end hears it. If I use a Cisco ATA with an analog phone and call the same person again

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,

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

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

Does anyone have the polycom soundpoint ip's successfully remotely provisioning? I've got the phone pulling default configs, and it's downloading phone specific information, but it's not actually using that information. Any help would be appreciated :) -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

I'm having some trouble with Polycom Soundpoint phones. I have had good luck deploying them on a local network, but now I've tried putting some in place which access their * server across the network. The * server is on a public IP and the polycoms are behind a NAT on a cable modem broadband connection. Every so often I get: May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com

Hi Everyone, 1- Are the Siren7 and Siren14 the G.722 HD voice codecs? 2- Are these codecs only for Polycom units or are they universal across all other SIP phones that advertise the HD voice codec like Aastra? 3- What is the main difference between the two and is it advisable to run these over the INTERnet (not INTRAnet)? Thanks -------------- next part -------------- An HTML attachment was

We've just setup our A@H server, with our quad port card. Everything works well so far. One thing I notice is that when I leave the handset on the hook and dial a #, all is well. If I pick up the phone and dial, it cuts off at 10 digits, which is a problem if I need to dial 1+area+phone # (12 digits). The phones are Poly Soundpoint IP 600's. I'm wondering if I've missed a

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

I am getting ready to install Asterisk and I was looking into the Polycom IP600 phones. I spoke with Polycom sales to verify the multiple line appearance and they said it would work. More specifically, if lines 1-3 all contain the same SIP registration info, the Polycom will only send out 1 SIP registration to the server and then handle the calls ringing on multiple lines. I was wondering if

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

Hello, We are planning to biuld a 100 lines PBX based on asterisk. How do you decide on the system config, e.i motherboard, cpu , how much ram , etc ? We will have all 100 phone plugged in. But we expect about 20 calls at any given moment. Thanks Varun

I've got a couple Shoreline IP phones, their Shoreline model number is Shoreline IP 100. I believe this is actually a Polycom SoundPoint IP 300 phone. I believe the phone is using a MGCP stack. I want to use it for testing with Asterisk. 1. I suspect I need to re-image the phone to make it work with *. 2. How can I preserve the current image on the phone? 3. What is preferred image to use

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi