similar to: ZYXEL wifi phones

---------- Original Message ---------------------------------- From: "Jim Van Eeckhoutte" <jim@vaneeckhoutte.com> Reply-To: <jim@vaneeckhoutte.com> Date: Mon, 8 Jul 2002 15:27:14 -0700 this is shorewall status output: tcp 6 431899 ESTABLISHED src=192.168.20.5 dst=64.4.12.45 sport=2185 dport=1863 src=64.4.12.45 dst=63.25.123.58 sport=1863 dport=2185 [ASSURED] use=1

hi,Jimmy Godbout when +86 13666666666 make a call to -> +86 020 87654321xxxx -> asterisk the CALLERID(num) will show the caller number +8613666666666 the ${EXTEN} is the dialed number 87654321 i will try the CALLERID(dnid) tomorrow, will this get the whole dialed number 87654321xxxx ? i don't know whether the sp will send me the whole number.... ------------------ ssmax

Asymmetric wrote: [SNIP] > > I don't know (don't use ices) but I thought I should inform you.. taking > user requests may cause you to violate the RIAA rules regarding that > topic, depending on how you handle it. Be careful. ;) > It's for my office... I'd love to play my CDs on my radio, but it would disturb those that don't like my music. Some do, so

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

ive tried your command and still not able to send thru outside smtp server . ISP is not blocking any ports. Also not able to telnet x.x.x.x 25.

Hi all I get a Cisco 1841 router, connecting the office through a CHAP PPPoE link: [...] interface Dialer1 ip address negotiated ip mtu 1492 ip nat outside ip virtual-reassembly encapsulation ppp ip tcp adjust-mss 1452 dialer pool 1 dialer-group 1 no cdp enable ppp authentication chap callin ppp chap hostname username at isp ppp chap password 0 s0meStr0ngP4sswd [...] As

Hi, I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver) to work with an asterisk box. The phone connects nicely to an external VoIP company (sipgate.de reportedly using asterisk themselves) but there is a strange problem with my asterisk: - Incoming calls via ISDN (chan_capi) to the Zyxel work perfectly. - Outgoing calls (capi/sip/iax) via asterisk are dialling out

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

(I posted this note on http://www.voip-info.org/wiki-ZyXEL+P2000W+configuration too) I tried to put together comments that were asked for on the P2000W. These configs seem to work fine for a ZyXEL P2000W, thanks to Giles Scott for getting me started with it. DTMF keys work fine and are "read" correctly by Asterisk. It's important that you upgrade your phone to a modern version

Here''s one I haven''t seen before I have a model "section.rb" as follows: class Section < ActiveRecord::Base ###Lot of common stuff end class Normal < Section end class Faq < Section has_many :faq_categories, :order => :position end class Rate < Section has_many :rate_tables end Now I''ve just added the Rates class and it works just

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 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 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,

I think we should add an FAQ entry for tcp_ecn. I remember Tom giving a good description in one of his many responses and there is mention of it in the pptp page, but I could not find the response from Tom about different tcp stacks. Thanks, -- Steve Herber herber@thing.com work: 206-261-0307 Systems Engineer, AMCIS, UoW home: 425-454-2399 ---------- Forwarded message ---------- Date: Sat,

Has anyone experiences this please: - We were running a number of ZyXEL P662HW-61 routers at our sites and all traffic was being sent over IP-SEC VPN's between devices. When we moved to a new architecture, we got rid of the VPN links so that the SIP traffic was running directly through the routers. Each site uses Snom 360 devices with the latest firmware (v4). Ever since we did this, the

Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only get 87654321, no area code . when someone dial +86 020 87654321xxxx , xxxx means 4 digits, the phone can call in, and the

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

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! I'm having a ZyXEL P2000W that I'm using together with my Asterisk box (CVS from some week ago). When I get a call directly to the *-box (lars@hostaname.domain.se) I see on the console that Asterisk get the calling users name as CIDName and his SIP-address as CIDNum - but at the P2000W i only get "asterisk" as CLIP? And I can see in the P2000W that "asterisk" is