similar to: Do I need a CH1 licence for Cisco Phones ?

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

I found a few mentions of the 7914 being used with Asterisk, these all covered SCCP/skinny though. Does anyone know if the 7914 can even be used with SIP? If so, any pointers? Is it a services thing? Anyone get the operator (line/extension status) to work with it. Thanks for the help, Cisco doesn't even mention ANYTHING about SIP + the 7914. Matt

Hi everybody, I will receive my CISCO 7960G tomorrow. I've ordered it as a "global spare" without any callmanager licence. Now I don't know if I can get firmware-updates so could please someone send me the SIP-firmware? Is the default firmware the "skinny" one? Wich would be better to use with asterisk? Thank you very much

I've been playing with Asterisk for a few weeks and it's working great. I have a question about getting multi-line receptionist phones working. I was thinking about getting one of these expansion ports: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet0 9186a008008883d.html What are people using for receptionist phones that show all the extensions in

Interesting notes on the 79xx series. The 7920 is the wireless phone; not mentioned here. For a more complete guide to Cisco's phones, see: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheets_list.html The 7902 is the "very inexpensive" Cisco phone, and it looks like it will be SCCP (Skinny) only. Twiddling my thumbs here waiting for the chan_sccp to

Dear R experts Sorry for posting: I have text file that I need to read into R (this is out from different program). A portion of data look like: ;example data 1_1 ch1 0.0 HAAAHHAAAHHAAAAAHHAHHAHAHHHAAHHAHAAAAHAAAHHAHHAAHAHHAAAAHAAAHHHAAAHHAHHHAAAHHHHAHHAAHAAAAHAHHHHHAAAHAHHAHHHHAHHHHHHHHHAAHAAHHHAHAAAAAHHHHHHAHHHHHAHHHH 1_2 ch1 0.1

Has anyone seen this issue with asterisk running like a dog when the internet is down ? Internal calls, incoming ISDN calls etc all seem to be affected. There is a local DNS server that is always available so I?m not sure why asterisk is so unresponsive. I?ve seen this on two different systems, and on 1 of them I commented out my SIP providers in sip.conf and it ran ok again. Thanks

Dear All, to my surprise as.factor does not accept a levels argument. Maybe I did not read the documentation well enough. See the example below. I wanted to use ch1 as factor in the newdata argument of survfit, so I assumed that I could write as.factor(ch1, levels=ch1), since the order should be kept. But as.factor(ch1, levels=ch1) results in the error: Error in as.factor(ch1, levels = ch1)

Hi, Please find attached the patches which I currently use on my desktop machine for dual head with the randr branch to fix the issues which I found. They may help others as well but may e.g. also disable the Xv blitter which might be working for some (but didn't on my card) - more information is found in the text comments in the patches. I have to hurry so this is short, will be back.

Does anyone have the 7970 work with sccp? I have used the example configurations from the wiki. I can see that the phone is failing to retrieve the CTLSEP<MAC>.tlv file and then is able to retrieve the SEP<MAC>.cnf.xml file. The phone is hanging with the message "updating ctl". Thanks, John SEP<MAC>.cnf.xml: <device> <devicePool>

I can buy a new ATA186 here, but it is sold with a 1-port user license UK, for euro 192, but does that license stop me from using both ports? I can't read the license agreement till I buy the thing, so I don't know what i'm buying... Michiel --

Well, I've asked a couple times already but I will ask one more, since I still havent' managed to get any further. We have several sites. Each site has it's own network and domain. We are connected via a WAN. Is the best practice to have one WINS server for each site? Or is a single WINS server for the WAN the way to go? Right now I have a single WINS server for the WAN. I CAN NOT

Full_Name: Kjetil Kjernsmo Version: 1.0.1 OS: osf1 Submission from: (NULL) (129.240.28.227) Ouch, another obscure bug... I just had a bus error crash, with a core dump, a small one this time. Usual <URL:http://www.astro.uio.no/~kjetikj/tmp/core>. I had done a few simulations, and done a chisq.test() on them. These simulations also write about 190MB's to file, but I have no idea whether

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

Can anyone tell me the features of the Cisco 7905 with SIP? I mean things like number of lines, speakerphone, transfer buttons, etc. I've seen the Cisco material, but all it told me was how nifty it is and how wonderful the XML interface will be ;) Thanks, --Ernest

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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

Try the following with R 63.3: test <- function(z){ ch1 <- deparse(z,width=500) ch <- NULL for(i in 1:length(ch1))ch <- paste(ch,ch1[i],collapse=" ") a <- gsub("(\\[(0|1|2|3|4|5|6|7|8|9|:|,)+\\])|([][+*/^():!<>%&|~,{}\"\\\=-])|( [0-9]+)|(^[0-9]+)"," ",ch) print("here") strsplit(a," ") } test(glm) Jim

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Dear R developers, I have a new class, which I called "Molecule", and have tried to define = a "+" operation for 2 objects of this class. This is what I have written so far, although the method is not complete = (I'm trying to look at it at intermediate stages): setMethod( f=3D"+", signature(x=3D"Molecule",y=3D"Molecule"),