similar to: Polycom Soundstation Conferencing Unit

> We run a variety of 5000, 6000, and 7000 series Soundstations running > Asterisk 11.6.0 and the phones are at 4.0.3.7562. We do not see these > registration issues. Would you be willing to send the configuration from asterisk for this? This message may be private and confidential. If you have received this message in error, please notify us and remove it from your system. Gyron may

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

I'm setting up a new Asterisk system on a Dell server and I'm getting "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk server. I've checked all of my codec settings and both the Asterisk and the Polycom agree on u-Law encoding. I'm using the latest release of the Asterisk code (1.4.17) and other software. If I call between phones (i.e. two

Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk? I can't see any mention of it on the wiki page: http://www.voip-info.org/wiki-Polycom+Phones Thanks, Alvin

asterisk-users-bounces at lists.digium.com wrote on 01/23/2015 10:24:24 AM: > Hello, > > I'm having a problem with a few Polycom SoundStation 6000s. > Everything works fine, but they drop registration to asterisk after > about maybe 30 minutes ? the phone does not re-try to register and > if you try to dial out on the phone it says ?URI Dialing is Disabled? > > Has

Hello, I'm having a problem with a few Polycom SoundStation 6000s. Everything works fine, but they drop registration to asterisk after about maybe 30 minutes - the phone does not re-try to register and if you try to dial out on the phone it says "URI Dialing is Disabled" Has anyone else had this issue? I'm running asterisk 11.7.0. This message may be private and confidential.

Hi, we've had a couple of Sounstation 4000's around for a couple of months working fine with our * box. Today, I tried for the first time to do a local 3-way conference with one of them, and could not find the "confrnc" soft key for doing that (as stated in the user manual). Spent 20 minutes without success, and ended up transferring the other parties to a meetme conference.

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

Just released by the CPSC on their recalls mailing list; please forward to any venues where you feel operators or resellers of the SoundStation might be, with this preface included. My 2W had a battery with the part code 1520-07804-002; its date code was GP0806, and therefore predates the recall period. Cheers, -- jra ----- Begin forwarded message ----- This message consists of the following:

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

Can anyone suggest a SIP phone that allows conferencing of more than 3 parties locally (i.e. not as part of a conference bridge)? I didn't realize Polycoms are limited to three. TIA, j

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty

I have a doubt... is it posible to do Video Conferencing using asterisk? -- Facundo Ameal. fameal<at>gmail<dot>com Linux User #395088 Open your mind, use open source.

Ok people it's been a while since I've heard from anyone about this. http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing The bounty for Asterisk Video Conferencing has now been bumped to $3,000. There are some additional feature and performance requirements for this funding now, please contact me if you are a developer who feel he has

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Gordon Henderson > Sent: Friday, February 09, 2007 9:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Conferencing Phones ... > > > Anyone got any experiences of good quality

I was wondering if anyone can help me with a problem we have at one of our sites. We have setup a Asterisk Trunk to a Avaya PBX, ie ... Avaya <-> Asterisk (1.2.30) <-> External ISDN Network BUT They also have a Polycom VSX 7000 that with some sort of BRI converters that plugs into the Avaya. The Trunk is working well except for Video Conference Calls. The Polocom can receive

Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference

Is there a good Web Conferencing add-on or a compatible package for Asterisk? I know there are web based controls for the audio, but I am looking for PowerPoint or desktop sharing functionality similar to WebEx. Anyone using a package they like?

Are there any speakerphones or other conferencing HW phones that play the audio in stereo? Either their own speakers, or jacks for an amp with room speakers? Is there any way for Asterisk to deliver call legs with stereo channels in the RTP stream? If not, is it possible for Asterisk to keep 2 separate calls, or pairs of legs in a conference call, synced exactly enough (including traveling over