similar to: newbie uk questions...

Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE

Hi, I'm interested in helping out with the Vorbis and Tarkin projects (in my already limited time!). I don't have an extensive background in all of the mathematics involved, but it looks like my mathematics from undergrad Physics is good enough to get a grasp on it. In the past few days I've started doing some reading on the mathematical basics. As part of my reading, I found

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

>But currently, I only have one ethernet jack per office. Routing >another 60 or so ports would add a very substantial expense in both >cabling and backbone expansion (what category ethernet is required, >BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is

On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: > For a very basic setup it would work, but I would suggest POE at a > minimum, and vlan support if possible. > > Gigabit uplinks, 10/100 for the poe ports > > http://www.amazon.com/NETGEAR-ProSAFE-M4100-D10-POE-Ethernet-Managed/dp/ B00AUEYX0Y/ref=sr_1_3?ie=UTF8&qid=1424462577&sr=8-3&keywords=netgear+poe

hi, My question is may be a bit out of scope but I don't know where to turn, I have a 1760 with a ccme 24 user licences 1 bri card. I want to configure a bri card in a cisco 1760 on port 0/0, the card is new, seen by the router, show isdn status gives layer 1 desactived , layer not activated, what ever I do, no shutdown command / shutdown command, etc , the green OK light never turn on,

Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't

I know the subject of faxing has been covered in some detail, but I was wondering if anyone has a hardware configuration similar to ours that has faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to Asterisk. We do not route calls over the Internet and our network has very low latency. The Asterisk

Hey has anyone seen a 1.6 cheatsheet around? Johnathan Snook did a nice 1.5 one but I''ve been working with 1.6 for a while and while I can use prototypejs.org, cheatsheets are handy for jogging memories.. I checked his blog, nothing there for 1.6. Gareth --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

Hola, Durante esta pasada Semana Santa, Santiago Mota y yo mismo hemos traducido al español la cheatsheet de "ggplot2" que recientemente publicó RStudio. Nos acaban de comunicar que la hoja de referencia ya está disponible: http://www.rstudio.com/resources/cheatsheets/ -- Saludos, Carlos Ortega www.qualityexcellence.es [[alternative HTML version deleted]]

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

You should be able to get ISDN though. I think BT provide this anywhere. Otherwise your solution will be a Primary rate (ISDN30) with 8 channels. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matt Sent: 07 June 2004 10:19 To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] Multiple DDI &

Dear Bredan -- I'm seeing them in the debug logs, but they're not making their way into the stream. OTOH, they are making their way to the directories (ie, shoutcast): http://yp.shoutcast.com/directory/?s=lamrim&st=gs&l=25 Any thoughts? Thanks for the great work! Roy At 03:38 PM 7/12/01 -0400, you wrote: >On Monday, 09 July 2001 at 12:26, Darrell Berry wrote:

I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a

Hello, I have just put reviewed versions of the "rsync cheatsheets" which I posted a couple of years ago (if I correctly keep track of time), right here: http://public.m-plify.net/rsync_cheatsheets/ - Sheet one lists the possible combinations of SOURCE and DEST arguments that may be given to rsync. - Sheet two lists the behaviour of rsync for different cases of SOURCE and DEST. -

Hi, I'm looking for opinions on the "best value" router to use for home offices. It should work for a scenario in which there are 3 computers and 2 SIP phones, handling QoS so that the phones always have higher priority traffic than the PCs. (and not rely on the phones to do the QoS because some PCs may not be connected to the phones). QoS could be based on destination and source

At the moment, we have a feature where if someone's sip extension is called, we also make another call to their mobile. We use the "c" option in the zap dialstring so that the user has to press "#" after answering to confirm the call (this prevents things like the answermachine grabbing the call if the mobile is switched off). We are now looking to move towards a sip

Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like