similar to: New User Help

I have a Cisco 7940G IP Phone. I am trying to load the firmware to SIP 3.2. The Phone just hangs in Defaulting CM to TFTP Server. It doesn't do anything else after that. I also have two other 7940g's that are the Universal Application Loader mode and say Protocol Application Invalid. I need to know what I can do to fix both these problems. I am running Asterisk@Home version 1.0. I have to

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

Hi, We've got a customer site with Cisco 7940G Phones connecting to our Asterisk Server via SIP. However, upon running a "sip show peers" it states that the phone is : user/user (Unspecified) D 255.255.255.255 0 Unmonitored All phones can call out via our Asterisk Server fine regardless. The problem is when they try and place a call to one another, where it

Is anyone here familiar with configuring Cisco routers? I have a Cisco 3620 with 3x WIC-1DSU-T1, 1x 2FE-2W, and 1x 1E-2W. I have 2 T1 lines being brought in by ACD.NET, a local CLEC. Within the next 3 months we will be getting another line from them, as our bandwidth needs grow. I am new to cisco and need help configuring this router. I am currently on a 1M DSL line with them. The DSL modem is

Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _____________________________________________________________________________ D?couvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et

Hello List, I'd like to ask how much you guys pay for an E1 (30 voice lines) and what company. You can email me personally and not the list. Best regards, Eddie _________________________________________________________________ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/

I'm looking at setting up Asterisk for a completely IP environment. All intercompany calls. I work for a ski area. I currently use a 3Com Superstack for in our office. And an old small town phone system for up at the mountain. The phone system is dying and I'm hoping to bring IP to replace the old phones. It will be about 100 phones at about 20 locations all within about 4 miles of each

Dear R People: I have the following in a file: resp factA factB 39.5 low B- 38.6 high B- 27.2 low B+ 24.6 high B+ 43.1 low B- 39.5 high B- 23.2 low B+ 24.2 high B+ 45.2 low B- 33.0 high B- 24.8 low B+ 22.2 high B+ and I construct the data frame: > collard.df <- read.table("collard.txt",header=TRUE) > collard.aov <- aov(resp~factA*factB,data=collard.df) >

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

Hello all, Here, we use rsync for years to share software tests distributions between from France to USA, Singapore and London, and all is working fine. But to save space we also need to compress files depending on their access time, and on purpose uncompress automatically before use. This makes rsync useless because it makes, of course re-transfer and deletions. when a new file is detected

-----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

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,

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

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

Hello, I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup. My plan is to use this handler to update my CDRs with values such as Asterish and Tech cause (see function HANGUP_CAUSE). I want to have my custom hangup-handler be run only once and when "the second channel" hangs up. At the moment, I'm issuing a couple of "channel originate Local/1 at mycontext1

Does anyone know of any sip wifi phones? Only one i can find that is redily availiable is the zyxel prestige 2000w and from what i hear it is flaky. Kurt Fankhauser WaveLinc HYPERLINK "http://www.wavelinc.com/"www.wavelinc.com 114 S. Walnut St. Bucyrus, OH 44820 419-562-6405 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database:

Hi, I have a couple of questions about MeetMe and call queues. I'm still pretty new to Asterisk, but already having to write a Service Center call manager for it (which I might add, our director has agreed to make open source!). MeetMe: How can I get MeetMe (does it even do this) to ask the user to speak their name first, and play that as the new member announcement. It seems

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