similar to: Polycom videoconferencing with asterisk?

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi, After upgrading to 1.5.2 I no longer can directly access to my voicemail by pressing the "Message" button, I have to go through the "urgent,new,old" report first. The oneTouchVoicemail parameter is set to 1 but not taken into account apparently. Anyone noticed that problem?

Hi Ulrich, <p>you distroyed my evening today ;-). Im only sitting here and follow the videoconferencing thread ;-)). > due to the ongoing discussion on both lists, i simply respond to both > lists. it's hard crossposting, but it's for both roups relevant (i > think). Yes, this was right. There is a great discussion and perhaps we got some points more yet. > +After

Anyone have any experience tying the Polycom VSX 7000e & Asterisk together? It says it supports standards based SIP servers but thought I'd see if anyone had real world experience. Thanks, Ken -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080303/8a7bbcc5/attachment.htm

Hello everybody, i am searching a solution for a videoconferencing, Any solution (Free/commercial). Asterisk is a great software, but recently we have more and more demands about videoconferencing of 3 or more peoples, Existing solutions are heavy and costly, around 2500? for 1 client. This is insane. Is there any solutions out there for non millionaires ? Or even Free ? I remember a company who

Speaking of video conferencing in conjunction with Speex, we are currently beta testing a solution we developed that offers multi-point, multi-party video and audio using the Speex engine for voice. http://www.wiredred.com/downloads/ecsetup.exe The fair and good audio settings are Speex narrowband, high quality is HawkVoice and toll quality is PCM. -----Original Message----- From:

I would like to participate also. Tony T. ----- Original Message ----- From: "J.K. Lin" <jk@pageshare.com> To: <speex-dev@xiph.org> Sent: Wednesday, November 19, 2003 12:29 PM Subject: RE: [speex-dev] Videoconferencing with speex and jabber <p>> Me, me, me, me too. > I would love to use it if there is one ;-) > (I wish I can contribute more, but Windows is

Hi all, <p>i have send a mail to the jabber mailing list and ask them how to send speex data with the jabber instant messaging protocol. I have added the mail adter this one. If someone here has any experience with jabber and speex please let me know. Thanks, <p><p>Carsten Breuer ====================================================== Hi all, im new on this list, so i want to

Me, me, me, me too. I would love to use it if there is one ;-) (I wish I can contribute more, but Windows is not my expertise :-( ) -- jk <p>> -----Original Message----- > From: owner-speex-dev@xiph.org [mailto:owner-speex-dev@xiph.org]On > Behalf Of Carsten Breuer > Sent: Wednesday, November 19, 2003 3:22 PM > To: speex-dev@xiph.org > Subject: Re: [speex-dev]

Hi all, <p>im new on the list and want to introduce my self. Im a software developer working on engine control software for a german car manufacturer. I want to use speex for a privat videoconferencing project and faces some problems with it. First of all the projects in Visual-C++ doesn't work. I don't mean the path an dependecy problems. That is easy. But there are some errors.

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest

> to speex: *now here comes the more important part, can we build a c++ > component which does what avrelay does? is it practicable to de/encode > 100 streams with a c/c++ speex de/encoder in realtime?* COMMENTS WELCOME At low bit-rate (6-8 kbps) and lowest complexity, it's probably possible to encode 100 streams on a 3 GHz machine (and decoding is cheap), but that's all

Hi Jean-Marc, <p>>>to speex: *now here comes the more important part, can we build a c++ >>component which does what avrelay does? is it practicable to de/encode >>100 streams with a c/c++ speex de/encoder in realtime?* COMMENTS WELCOME > At low bit-rate (6-8 kbps) and lowest complexity, it's probably possible > to encode 100 streams on a 3 GHz machine (and

Hello, Our Polycom's 601 can no longer register or communicate with the asterisk server when using kernel 2.6.18.x. Cisco 79XX and other phones still work though. Downgrading back to latest 2.6.17.x solves the problem for Polycoms, but I'd really like to understand what's going on there... Any idea?

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

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to