similar to: ZyXEL 2000w In Call Menu/Hold configs

Hi, Is it possible to configure NTLM HTTP proxy authentication using the winbind/squid "ntlm_auth" helper, to authenticate users against a Samba 3 server? I already have the NTLM auth working against a Windows 2003 Active Directory, but I also have a completely separate Samba 3 server that I would also like to configure NTLM proxy authentication against. Please advise, as I can't

Hello I have just spent a happy day (!) configuring Exim4 to accept authenticated connections from full blown Outlook clients. That works. Unfortunately, by ticking the "Use SPA" box in Outlook it now demands SPA on the POP3 server as well. The existing POP3 server doesn't support this style of authentication. I believe Dovecot will. We're already using it as an IMAP

Hi, I''m wondering how Shorewall will behave if I want to use it on a router with some if its interfaces in a bridge, and some not in a bridge. For instance, eth0 and eth1 bridged as br0 and eth2 as a stand alone interface. Dan... -- Dan Searle Adelix Ltd dan.searle@adelix.com web: www.adelix.com tel: 0845 230 9590 / fax: 0845 230 9591 / support: 0845 230 9592 snail: The Old Post

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Hi, I know we've talked about this phone to death. I have pretty good voice quality, with and without WEP enabled, using the G729a codec and DLink & Netgear access points. I am facing one obstacle that is driving me insane. Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold,

Hi All, Today I got a couple of ZyXEL Prestige 2000W WiFi phones. I'm having a problem making SIP calls although I can receive calls just fine. When I try to make a call the phone makes some sound (like "bup bup bup bup bup bup beep beep") and then I just hear hissing background noise (not too loud - like comfort noise). I upgraded to the latest firmware on the phone - Wj.00.10

Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) What can I do to debug this in the CLI. Nothing seems to shed any light on the problem. I am using

Hi; I have entries as below in DB, mysql> select * from sip_buddies; +----+------+----------+------------+---------+------------+--------+------- -----+------------+----------+------+ | id | name | context | defaultip | host | mailbox | type | regseconds | ipaddr | username | port | +----+------+----------+------------+---------+------------+--------+-------

I've just seen this post: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg41132.html and it took me back to play again with my dust collecting 2000W. Does anybody got DTMF to work? My sip.conf looks like this: [400] type=friend context=from-sip username=400 secret=verysecret disallow=all allow=g729 dtmfmode=rfc2833 host=dynamic nat=yes qualify=300 canreinvite=no My phone is

Hi, Has anyone tried the ZyXEL Prestige 2000W with * ? Is it worth the money ? Best regards Matthias -- _;\_ Matthias Cramer / mc322-ripe System & Network Manager /_. \ Dolphins Network Systems AG Phone +41-1-847'45'45 |/ -\ .) Libernstrasse 24 Fax +41-1-847'45'49 -'^`- \; CH-8112 Otelfingen

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety." Benjamin

Does anybody have any experience with the ZyXEL Prestige 2000W? I am having problems with the line tear down when I call another extension. If nobody picks up at the other end when I hangup the 2000W, the other extension continues to ring. Is there any way to hangup a SIP call if there is no more traffic? Asterisk seems to think that there is still a connection open. This is pretty annoying

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

No, it will dial like a pass-through simultaneously to sip/iax extensions. If you were to dial out to an analog port though, that would be different. So in essence, you can have all the phones ringing at the same time. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian J. Murrell Sent: Thursday,