similar to: Cisco POS 3-08-2

Hello, I have a problem with creating an identity matrix for glmnet by using the contrasts function. I have a factor with 4 levels. When I create dummy variables I think there should be n-1 variables (in this case 3) - so that the contrasts would be against the baseline level. This is also what is written in the help file for 'contrasts'. The problem is that the function

Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -

Hi all, i have a problem with some cisco 7960. Yesterday i did a firmware-upgrade from 3.1 (1.2) with P0S30203.bin as described in the most documents. Now i get the message Phone unprovisioned and in TFTP-Log i find the following line: 07.03.2005 19:58 :Timeout error sending P0S3-07-3-00.bin to 192.168.0.83, 0 bytes So i think it now wants to load the file P0S3-07-3-00.bin. But i have not

I know this isn't a specifically asterisk question, but does anyone know how to make the phone NOT use it's old config? I'm trying to get rid of the second line registration crap and it's not working. Aaron

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031207/e5e9b8eb/attachment.htm -------------- next part -------------- Hello all, I am newbie to Telephony world (IP and PSTN). Please excuse me if you find my questions very dumb. I am trying to configure my IP 7940 with the Asterisk, when phone boots up it only shows the message "Phone

Is there a way to have extensions automatically created for registered sip users ? I did some investigation and found some hope in chan_sip with relation to the somewhat undocumented usereqphone option but i may be totally off track. All i want to be able to do is send a call to number@ip_address where the number is the username configured on the phone that has registered with asterisk

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others

Folks, I have CISCO 7940g phone. I have in the past configured the phone with two lines. Having found the 2nd line wasn't much use, I want to remove it from the config. I have taken it out of the SIP config file that is TFTPd to the phone but it is still showing on the phone and it is still trying to log into Asterisk with that account. I have tried removing the config line and blanking

I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed "/DEFAULT_CIDRINGS 2"/ Is there something i'm missing ? My Cisco 7960 just shows "asterisk" Thanks, Nathan [zapata.conf] context=incoming usecallingpres=yes relaxdtmf=no rxgain=0.0 txgain=0.0 busydetect=no pridialplan=local usecallerid=yes callerid=asreceived

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them.

Is there a reason the variable ${DNIS} does not get set with incoming calls via chan_capi ? Is it related to the MSN=X in capi.conf ? version = chan_capi-cm-0.6.3 example; exten => _95555555XX,1,NoOp, ${EXTEN}, ${DNIS} == ISDN1: Incoming call '0400000000' -> '95555555' -- Executing SetCDRUserField("CAPI/ISDN1/955555 55-135", "Incoming") in new

I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960 and 7912 currently connected and functioning. I'm trying to use the recommendations from here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP I have created a "XMLDefault.cnf.xml" and it took the latest image but the phone states it's unprovisioned? Any

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

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxxxxxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and reset the SIP parameters. How do I make it "forget" what it has programmed and

I've just recieved a copy of the new SIP firmware for the Cisco 7970, those of you with Cisco accounts may wish to try it (shock horror I'm sticking with SCCP). This coincides with the release of v8 firmware for all Cisco phones (and for those of you running Sergio's chan_sccp v8 works fine) The firmware is now also (and for the 7970 SIP, only) distributed in ".cop" files,

Thanks for the feedback. I did some research and it looks like you were calling over international lines. It also appears that there was high than average static on the line, which is not normal for my system. It's true that I threw my recordings together quickly and the beep was supposed to be funny - it was actually me saying "beeeeeep". However, the static and noise you received

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

Hi all, I have seen similar discussions about this problem earlier, but I need some help here! I've been using this phone for allmost two years without any problems. Just about a week ago I had the phone unplugged for a few days and when I plugged it in again it had lost all settings, including the settings password. It was reset back to the factory default. Well, then I decided to enter