similar to: Re: asterisk-users Digest, Vol 31, Issue 37

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Gordon Henderson > Sent: Friday, February 09, 2007 9:47 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Conferencing Phones ... > > > Anyone got any experiences of good quality

On Thursday 08 February 2007 07:32, asterisk-users-request@lists.digium.com wrote: > Does anyone have any recommendations for a phone that has easy to > understand/implement central provisioning? I've used CISCO 79XX phones, > and they're great (but too expensive). I like Grandstream phones, but > their provisioning sucks. > > What is everybody else using in large

Greetings list, Does anyone have experience with SIP conference phones? I need to source a couple for a client, but I'm not really familiar with the market - i.e. what's available, what's decent quality, etc.. A cursory googling has led me to the Polycom Soundpoint IP4000 at around the ?450 mark - any thoughts on this? If anyone knows a good Polycom wholesaler in the UK, I'd be

Greetings, (Apologies if this is an FAQ, but I've Googled for hours and haven't come up with anything yet.) I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This

I'm trying to get a few Polycom IP4000s working with asterisk, the incoming calls from inside and outside of the network work fine with it, but when I try calling out it just kicks me over to a busy siginal. Anyone have any ideas on what causes this or how to fix it? Thanks, Nathan -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi, Most embarrased newbie evere here again. Possibly another daft question. I have the digium starter kit lite, so I've got the single FXO and FXS lines All is working well with local sip phones able to dial other phones, conferencing, MOH (Thanks Asterisrk-users list!) along with the one analogue handset etc etc. The one niggleing problem I have now is this: My Dialplan is set to ring

I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on a flat local network. I followed the provisioning guides that I found on the Web, and I have the phone downloading bootrom.ld, sip.ld, and a bunch of configuration files. This all works properly. However, I receive the following error: NOTICE[27345]: chan_sip.c:14725 handle_request_register: Registration from

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

Greetings, We have An Adtran 616 Total Access device talking to a colocated Asterisk machine over MGCP. Calls placed to the phones connected to the Adtran go through as do outgoing calls from the phone (prefixed by 9), but feature access codes (*97 for voicemail, for example) and extension-to-extension calls don't work. As soon as the first digit is pressed, the user hears a busy signal.

Hi, I am looking to integrate Asterisk with a Siemens HiCom 150e via BRI and wondered if anyone is able to offer any advice. In simplistic terms, my goal is to pass calls from the HiCom to the Asterisk box. e.g: HiCom user dials access code and can call Asterisk extension or establish SIP call over Internet. Likewise, I'd like Asterisk to be able to present a call to the Hicom, either

Greetings, We are trying to use Samba 3.0.10 running on FreeBSD 5.3 to replace a legacy NT4 PDC. Our goal is to use LDAP to centralize all user information and authentication on the network. To that end, we've set up Samba to use LDAP for authentication of all the Windows users. This is working, but Samba seems to require that all Windows account have a matching Unix account as well. This

I am having problems getting incoming calls from the PSTN to route to extensions, digital receptionist and even voicemail. When I call a DID number for one of the lines, it rings twice then says: "Goodbye" and hangs up. (logs to follow below configuration info). When I dial 7777 it goes to the digital receptionist without any problems. The system setup is simple; I have 8 PSTN

Hello, I need adddriver to samba network printer, but I have problem with too many driver depend files in it. String with files is 1928 char long I am receiving this message: ERROR: string overflow by 1 (1024 - 1023) in safe_strcpy [adddriver "Windows NT x86" "Canon PIXMA iP4000":CN] this is my attempt: rpcclient -U 'admin%passwd' -c 'adddriver "Windows NT

I sent a query about this before, but have some further information and am hoping somebody has a suggestion as to what to try next to debug this. I'm using an Asterisk box primarily for MeetMe conferencing. There are two sources: TDM via two Q.SIG T1's and SIP phones. Conferencing works fine between TDM channels. But when a SIP phone calls the conference, there's no voice path *to*

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

I have the same problem. Thanks for fix it. r-sig-debian-request@r-project.org escribió: > Send R-SIG-Debian mailing list submissions to > r-sig-debian@r-project.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://stat.ethz.ch/mailman/listinfo/r-sig-debian > or, via email, send a message with subject or body 'help' to >

Just came by this link So I'm posting to keep the community informed. I don't use or endorse this product. I'm just letting people know about it. http://www.indosoft.ca/audioconferencesystem.htm Audio Conferencing System & Teleconferencing Solution that connect seamlessly over TDM and IP networks. This audio conference system include a comprehensive set of features and easily

Can someone please help me with my config settings for my TDM12B. I have got the FXS port to work fine, but I cannot get asterisk to startup with any settings for the fxo ports. It appears to be something related to the message "Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling". However, I have tried what appears to be every possible configuration option. I

Has anyone here ever actually truly successfully gotten a Polycom IP7000 to take a productivity suite license and enabled the bonus features like 4-way calling, recording etc? It ALWAYS works perfectly with ALL of our Soundpoint IP 5/6xx phones, but NEVER for our IP7000s. I just want to know it's POSSIBLE before I keep slogging away at this. Is there a 'bastard_phone=yes'

Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I