search for: ip650

Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the IP650. When a call is placed between those two the au...

Does anyone have an actual delivery date on the new Polycom HD IP650's? I'm getting sick of not having a backlit screen and thinking of upgrading. Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061018/9937184f/attachment.ht...

Anyone seen anything on the IP670 & the Color Expansion? -Matt

On the polycom soundpoint ip 650 six line phone: Say I have 4 lines on hold, is there way to tell who I put on hold. I cannot see the caller ID of the other lines, only the last line I placed on hold. Thanks, --E -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris

So I've got a bit of a head scratcher. Wanted to get some insight. I've got a PBX running 12.3.0 We're a ULAW shop from end to end. But I've been playing with G722 just for fun. I've got a Yealink T46G on my desk, And my colleague, A Polycom IP650 (Same office). Basically, Whenever I make an outbound call to a destination to something not G722 ready, I get no incoming audio. Example 1. From my T46G, I call the IP650. The call comes up in G722 and we can both talk, No issues. Example 2. From My T46G I call outbound to my cell ph...

> I ran into this problem with an early batch of IP650s. Polycom's firmware > version 2.0.3b made this issue go away. Speaking of Polycom firmware, anyone have an up to date source for the stuff? The site I ordered from took down their FTP site that had it. :( -Kenneth

I am trying to get something working that is just not doing quite what I want. It may not be possible, but I figured it was worth asking about. The details: Asterisk 11.6.0 Polycom SoundPoint IP650 phones running 4.03 firmware. We have a queue with 4 phones in it. ringinuse is set to yes and the stategy is ringall. In sip.conf, we have notifyringing set to yes as well. Asterisk is sending messages of the type application/dialog-info+xml to the phones. This works nicely in almost every sc...

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

In a recent interview someone from Digum indicated that the G722 wideband codec was being worked into Asterisk. This will make Asterisk compatible with Polycom's new HDVoice products like the IP650 phone. This is very interesting, potentially exciting, but it brings up certain questions. Who will benefit as long as calls must typically pass into existing PSTN infrstructure, and so be transcoded into G.711? It seems to me that only systems that are IP end-to-end stand to show the improvement...

I'm looking for some ideas of people who have setup a sidecar (eg: Aastra 560M). Obviously it's handy for BLF (to see who's on a call)...but what else? Anyone want to share interesting things they've done with a sidecar? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'd like to disable MWI on certain lines of my IP650 Polycom phone. So I removed the "mailbox=" parameter from that line's peer section in sip.conf. Yet the envelope still appears in front of that line and the phone MWI keeps blinking. Where should I look to completely disable MWI on a certain line? Thanks,

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 =>...

Hello, Several people report that outgoing phone calls to our clients sound muffled, like they are talking underwater. Reported for both the Snom 870, and the polycom ip650. Incoming calls sound ok. Could this be a codec problem? My dialplan looks like: [general] port = 5060 bindaddr = 0.0.0.0 srvlookup = no tos_sip = cs7 tos_audio = ef registertimeout = 1 relaxdtmf = yes context = testofidea disallow = all ;allow=gsm allow = ulaw allow = alaw allow = g722 dtmfmo...

...if (strstr(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 > exten...

I'm going to be rolling out asterisk at a small office and one requested feature was the ability to have a phone that can be configured so that ringtones can be configured according to the callerid of the caller. Does anyone have Asterisk experience with such a phone? Any suggestions would be greatly appreciated. Thanks in advance!!!

Hi, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 650 Data: ? P/N: 3150-11530-212 ? SD Sound ? FW: 2.1.2.0078 ? Assembly:

...ailto:jeff at 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> ex...