similar to: Polycom 550 IP SoundStation Fuzzy Voice Quality

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?

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.

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

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

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

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

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

> 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:

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

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:

I've searched high and lo and googled to I can't google no more... I knew that Cisco bought Selsius to get their VoIP solution but what I didn't know was that the 12sp+ is based upon an ITE-12 product that is apparently used at universities. I've taken two of the phones apart and started swapping parts and have looked up the IC's on the Internet... Anyway... After using

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

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

All, Can you use the global variables set in the [globals] section of the extension.conf file in other configuration files? Here is an example: I have a file called globalvars.conf that I include in the [globals] section in extension.conf. I would like either (a) that the global variables being defined in this section can be used from voicemail.conf or (b) that I can include the

I need some help summarizing complex data frames (small example below): m1_1 m2_1 m3_1 m1_2 m2_2 m3_2 i1 1 1 1 2 2 2 i1 2 1 1 2 2 2 i2 2 2 1 2 2 2 For an arbitrary number of columns (say m1 ?. m199) where the column names have variable patterns, and such that each set of columns is repeated (with potentially unique data) an

I have a polycom 501 phone that is 1 hour off now. Before last sunday (time change) the time was fine. <?xml version="1.0" standalone="yes"?> <PHONE_CONFIG> <OVERRIDES _.0x20._log.level.change.sip="0" tcpIpApp.sntp.daylightSavings.stop.date="4" tcpIpApp.sntp.daylightSavings.stop.month="11"

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