search for: ip335

I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with 3.2.4 it would automatically show call waiting name and number without pressing any keys. It could be

I am posting here since you guys are my last hope. I am trying to configure a Polycom Soundpoint IP 335 with MWI. Is there any way to eliminate the scrolling messages and Msgs softkey? I am trying to get it where it's just the light that indicates the new messages. I don't know if Asterisk has to send a different notification or what have you. Thanks, --Eric -------------- next

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 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 th...

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same firmware as we...

....me/next (12 Noon EST) why not join us on IRC: #vuc on Freenode.net anytime SIP:200901 at login.zipdx.com Skype:vuc.me (or skypeld.vuc.me for reduced bandwidth) PSTN: (567) 252-2286 Java web widget: http://vuc.me/call If you have a shipping address in North America, you can vie for a free Polycom ip335 : http://bit.ly/8px6al Independent of your location in the world, Howler Technologies is offering some free licenses for their g729 transcoding technology. See you there! /r

...loading these for $32 Each - Buy one or 100 :) > > I'll also entertain offers for bulk orders. > > http://short.e4strategies.com/dph540 > > Kind regards, > > Mike White > .e4 > http://8774e4voip.com > > PS - I also have many new Aastra 6739i and Polycom IP335 phones that are > available at substantial discounts. > Reseller? Contact me... > > > > > > -- > __________________________________________________________________ > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk...

Struggling with an IP650 and 7 IP335s this morning. I have the following hints defined (courtesy of FreePBX 2.9): extensions_additional.conf:exten => 300,hint,SIP/300 extensions_additional.conf:exten => 301,hint,SIP/301 extensions_additional.conf:exten => 302,hint,SIP/302 extensions_additional.conf:exten => 303,hint,SIP...

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

Greetings, I mailed the list regarding an intercom system some months ago and we came to the conclusion that I should purchase a Polycom 501 phone. I'm now considering the purchase for this year, and I'm now wondering between the Polycom 501 and the 320 for the intercom. I don't need the spare ethernet on the phone because I would like to have my voice network separate from my

...r(p->useragent, "Polycom")) { p->subscribed = XPIDF_XML; and change that line to p->subscribed = DIALOG_INFO_XML; On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere <jeff at sunfone.com> wrote: > > Struggling with an IP650 and 7 IP335s this morning. I have the following > hints defined (courtesy of FreePBX 2.9): > > extensions_additional.conf:**exten => 300,hint,SIP/300 > extensions_additional.conf:**exten => 301,hint,SIP/301 > extensions_additional.conf:**exten => 302,hint,SIP/302 > extensions_additi...

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...t sunfone.com" target= =3D"_blank">jeff at sunfone.com</a>&gt;</span> wrote:<br><blockquote class=3D"= gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-= left:1ex"> <br> Struggling with an IP650 and 7 IP335s this morning. =A0I have the following= hints defined (courtesy of FreePBX 2.9):<br> <br> extensions_additional.conf:<u></u>exten =3D&gt; 300,hint,SIP/300<br> extensions_additional.conf:<u></u>exten =3D&gt; 301,hint,SIP/301<br> extensions_add...