similar to: RECALL: Lithium batteries for Polycom Soundstation 2W

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

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?

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.

> 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, 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.

Does anyone know if you can use the Polycom "Norstar Clarity" speakerphones with Asterisk? Model number is 2501-03308-001 'C' Is it a SIP handset or analog style unit (or worse proprietary). Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

Hello, Recently, the following problem started happening with a particular samba server: If i have a file open for reading (say, a pdf in xpdf) and then try to write to it (say, through recompiling a latex document) it complains that it cannot open the file for writing. this seems like a file lock issue but I am unsure where it is happening. My previous usage should be perfectly safe since

Hi All, I tried combing through the archives, but I think that I just went mental. We're trying to set up a pxe server here using syslinux 2.0.6, and here's what we've got: client sends bootp request receives ip, server to go to and file to fetch. fetches and executes pxelinux.0 fetches pxelinux.cfg/default fetches display.msg (defined in default) displays the text - but when I

On Tue, 24 Jul 2007 wilsonk at cpsc.ucalgary.ca wrote: Hi Kelly, Sorry for the delay, I'm way behind in email :( > I built llvm-2.0 (the svn version) by just typing "./configure;make". I > then downloaded clang from the svn repository (revision 40382) and changed > to it's directory under .../llvm/tools/clang. I did a "make" without > incident for clang,

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After 2 good updates to 4.4, my 3rd is giving me trouble. I did all 3 installs the same: yum clean all yum update python-sqlite sqlite yum yum clean all yum upgrade This one, however, quits during the header download. I get "[Errno -1] Header not complete, trying another mirror", then "[Errno 256] No more mirrors to try". I am not going thru a proxy, and I can't find

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

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

I would like to know if any body is using the Polycom Soundstation IP 4000 SIP conference phone with Asterisk. I am thinking of purchasing one. Kurt

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

I have InAccess Networks' oh323 installed and partially working. I can call the h.323 phone from asterisk using Dial(oh323/${IP_ADDRESS}). How do I dial from the phone to an asterisk extension? It does not appear to me that the phone actually registers (or attempts to register) with asterisk. I'm using Asterisk Stable and the phone in question is a polycom Soundstation IP 3000 or