similar to: Conferencing Phones ...

I am perplexed by this so I how someone can help me out. On one of my servers the users began complaining that if they picked up a parked call they could not use the # key to transfer the call. This is a particualarly annoying issue since everyone has been taught to use #700 to park calls. At first I thought it was a DTMF issue with the polycom phones, since rebooting seemed to fix the problem.

Hey all, I have a situation where I have 8 lines from the phone company in a hunt group coming in to my asterisk box. These are the same lines I'm using for outgoing calls ( named g0 ). The problem arises when someone dials our number at the same time asterisk tries to put a call out on one of the zap channels in the g0 group. This has happened twice that I know of so far, once to

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

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

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

Jay, I have had a lot of trouble with the FXO ports on Adtran TA750. Unless the incoming POTS lines have a balance impedance, they will buzz very bad. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jay Milk Sent: Sunday, April

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

Run /usr/src/zaptel/ztmonitor 32 -v And adjust your gains in /etc/asterisk/zapata.conf accordingly. Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of bam Sent: Friday, May 07, 2004 3:35 AM To: asterisk-users@lists.digium.com

On Friday 09 February 2007 11:50, asterisk-users-request@lists.digium.com wrote: > Anyone got any experiences of good quality VoIP conferencing phones? > > I've used Polycom analogue units in the past, and I see that they have a > SIP version (the IP4000) - but it is better/worse/as good as an analogue > version? > > (ie. would I be better off with an analogue version

All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank

Hi All I have a snom 200 phone here which works perfectly when using the handset to playback the voicemail messages etc. However when I play back the voice using the speakerphone it sounds choppy. Anyone had this problem before? Regards Mark

Greetings, I have a new * system installed and everything works as it should except for one annoying little problem: I can't play any sound files. What this means is that when an extension script gets to the point where it should play a sound file (voicemail greeting, auto-attendant, whatever), the caller hears a click and then silence. According to the * log, the sound file is being

I'm running Asterisk CVS-HEAD-05/24/04-17:37:48 on kernel 2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here are the pertinent files: zaptel.conf: fxsks=1-4 loadzone = us defaultzone=us zapata.conf: [channels] context=north_in_pots_vip group=1 signalling=fxs_ks usecallerid=no hidecallerid=no callwaiting=no restrictcid=no threewaycalling=no echocancel=1

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

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'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, 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?

I need a good reliable source for channel banks. Ebay is nice but too many variations of quality. I need one that can do 8 FXO and 16 FxS does anyone have a particular model, other than adtran that they can recommend??? And if possible include a source to buy from... Thanks in advance, Alex -------------- 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