similar to: ZyXEL Prestige 2000W

Hi All Just got a new quadBRI card and connected one port to our Old PBX. When I make a call from a sip phone to a phone number the phone rings, I hook up, and the call on the sip phone allmost imidialely disconnects, after a fiew seconds the "real" phone disconnects too. Here is a trace: -- Executing SetCallerID("SIP/cramer1-b718", "45") in new stack --

Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3

I am just an Asterisk newbie doing a test install. I am using 2 X-Lite clients and have configured them according to the wiki on voip-info. A warning is still displayed on the Asterisk server console saying that I should disable RFC3389 on the client, even after I changed the Transmit Silence to yes. I am able to connect and call the other client, but when I do no audio is being transmitted by

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was

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 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

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 all I have just got a P2000w and experience several problems. Hopefully there is someone out there that has got it working. I saw it on Cebit and the person demonstrating it there told me that it was connected to an Asterisk server on the stand -so it should work. Problem 1: it does not register correctly It get lots of messages like this: Jun 26 19:45:19 NOTICE[1107585968]: chan_sip.c:5630

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 Everyone, After a fair amount of faffing ive managed to get the 2000w working with asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls over our BT line). The final solution is to set up outgoing VoIP calls but I now know that without a SIP aware router I can think again! (damn you iptables!) In the mean time I'm trying to figure out why I can't get the

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, Does anyone have suggestions hooking Zyxel Prestige 2002 to Asterisk? I have tested Zyxel Prestige with both supported codecs. Call with G.711 sounds very choppy and cracking. Almost can't understand a word. Today I installed G.729 support into Asterisk but unbearable voice quality remains. It's a little bit better though. I have tested that Zyxel ATA with some commercial SIP

Hi Has anybody tried to use the Zyxel Prestige 2000W_v2 with Asterisk? I have the latest firmware and can now make outbound calls. So it looks like Asterisk does accept the configuration in sip.conf. However, I cannot receive any calls. Also, I keep getting this message on the console: chan_sip.c:7733 handle_request: Registration from '<sip:51@10.99.1.133;user=phone>' failed for

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

Hello Everyone, I am using GlusterFS 9.4, and whenever we use the systemctl command to stop the Gluster server, it leaves many Gluster processes running. So, I just want to check how to shut down the Gluster server in a graceful manner. Is there any specific sequence or trick I need to follow? Currently, I am using the following command: [root at master2 ~]# systemctl stop glusterd.service

Greetings: I have a working configuration for SCCP on our LANS which doesn't route RTP correctly to a skinny phone behind NAT registering from a remote public IP. Configuration: asterisk 1.4.35 servicing only skinny phones trunked to asterisk 1.2.40 which services chan_phone FXS, zap FXO and SIP phones; both instances of asterisk are behind NAT and run on the same host (using different base

I see that some, but not all, releases provide a local link to the key used to generate the signature files, which makes it difficult for a script to use them to verify the signatures. Gcc solves this problem by including the following on their mirrors page ( https://gcc.gnu.org/mirrors.html): The archives there will be signed by one of the following GnuPG keys: - 1024D/745C015A 1999-11-09

(I saw this flipping through the archives for something else.) Marek, did the bcmxcp driver work for you or is your UPS too old? John