similar to: Polycom distributor in the UK ?

Hi; I've got an AAH installation where a customer wants to install an active Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel 2.4.21.37. Support for Eicon active cards is built-in. I've debugged and run the A@H install-Eicondiva script but when I try to run divactrl load -c 1 -f ETSI -Debug I get a response : A: can't get card type for DIVA adapter

I have an Asterisk box running happily under Fedora Core 2 with a X100P and a TDM400P, and now I'd like to integrate it to my ISDN2e connection using either an AVM Fritz PCI card or an Eicon DIVA passive card, both of which I have sculling around. I've successfully used the AVM card under RedHat 8.0, but I'm having difficulty finding information on running it under the Fedora 2.6

Hi; I've been asked to take a (remote) look at an Asterisk@home system running asterisk 1.0.9 on Centos 3.5. It's running chan_capi-0.3.5 It has an AVM c2 ISDN card which is plugged into what I believe to be a couple of BT ISDN2e "System Access" (i.e. point to point) connections. We've placed a support call to BT to find out how these lines are actually provisioned, but

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Jerry Glomph Black > Sent: Monday, January 30, 2006 11:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Polycom IP301: Pass-through ethernet port > unusable? > > Have just done a

Hi, I am about to buy ip pbx asterisk system but what ip phones do you recommend? Are polycom ip all functional with the ip pbx system??? Be waiting.thanks a lot Marlo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050811/16cc52cd/attachment.htm

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

We have tried Asterisk 1.0.9 on FC4 and have never been able to get CAPI (with Fritz card, fcpci) to work properly. Apart from that Asterisk works fine in switching internal calls. But it's useless if we can't make outgoing calls on our ISDN line. We are considering abandoning FC4 for Debian or SuSe. What is the general concensus on the best Linux to run Asterisk with CAPI? /Why Tea

Hello, We have a problem on a recent asterisk install with Polycom 30x phones; Sometimes (can not reproduce or find the logic of the problem after one week one analysis), the called party (even incoming or outgoing call) can not hear the calling party, as other flow works (caller hears called). This occurs between 5 and 10% of the time. The configuration is the following: - Asterisk 1.2.9.1 -

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Yeah....I think that every install I have done the first thing that happens is "why is there a delay before the call connects?" and the answer is "you have to hit dial or wait 10 seconds". -Jonathan -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Tom Rymes Sent: Thursday, August 11, 2005

This feature also works on the IP301 phones. The obvious caveat is that it is one-way only. Still nice for an "all-page" though. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith > Sent: Thursday, February 09, 2006 12:27 PM > To:

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks

My polycom phone is now hanging at Running "sip.ld". I modified it's config via the web interface to register with my asterisk box. I have tried to restore the default settings wth 468* and it doesn't seem to work. Any ideas? -jonathan

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Hi all, Anyone using their services ? I'm thinking of setting up my servers with their service. But before starting to mess with my extensions.conf I thought "let's check the community for their experience". Thanks, Michiel van Baak.

Hello, I am looking at the Polycom phones. The ip600 has a very nice screen, is that the only real advantage over the ip500 and ip300.. Is it worth the extra dollars? Kind regards Michael Felder IT Medic Australia Pty. Ltd. P: 03 9557 2213 F: 03 9557 2214 M: 0419 568 217 E: Michael@ITMedic.com.au <blocked::mailto:Michael@ITMedic.com.au> http://www.ITMedic.com.au

Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad

Hello I have the following strage problem here. I have the following smb.conf: ---cut--- tux:/shr/pdfdropbox# more /etc/samba/smb.conf [global] workgroup = de_zycko netbios name = tux server string = tux kernel oplocks = No encrypt passwords = Yes guest account = Nobody invalid users = root # This tells samba to write log files per

We've got a Toshiba DK system w/ analog ports that went to a voicemail server. I swapped in an Asterisk box with a Digium 4-port fxo card. It /almost/ worked perfectly. The problem is that Zap channels never hang up. They have to time out. I set up MeetMe, but all Zap channels hung forever. Very annoying. Same thing for FXO-to-FXO bridges. I figured out today why and fixed it.

I have been testing with Polycom IP600 phones for a month or so. I found out that there are frequent problems with the handset. The problem is I can hear the other end but the other end cannot hear me. I have already downloaded the latest bootROM 3.1.3 and sip 2.1.2 However, there are no problems with the headset or speaker phone. Has anyone encountered such problems before? Thanks.