search for: 7941

Anyone aware of how to configure one call per line on a Cisco 7941/7961? The default behaviour is to have two calls per line button, and this is confusing for some of my users so I'd like to be able to have the 2nd call ring the second line button, rather than being shared with the first. I'm hoping this is something that is configurable in the XML or on t...

Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy at manistee.org -------------- next part ----...

I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEP<mac>.xml there is a section called "softKeyFile" where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this f...

https://bugzilla.samba.org/show_bug.cgi?id=7941 Summary: Formatting bug in rsync man page? (man -k, whatis, catman -w) Product: rsync Version: 3.0.7 Platform: Sparc OS/Version: Solaris Status: NEW Severity: minor Priority: P3 Componen...

Just saw this on Cisco's software download site: 7941/61 IP Phone SIP phone load - for CCM v5.0 Has anyone used this with Asterisk yet? Josh

I'm playing with phone ci$co 7941 with sip image (8.02SR1), strange is, that phone displays caller id number with ip address of asterisk server like "8210@172.20.24.11" I think, this is some bug in firmware, but I would like to find some workaround, maybe using SIP_HEADER function, but seems, that this can be used only...

Hi all, I have a bunch of Cisco 7941's with SIP firmware 8.2.1 that work great on a LAN with Asterisk 1.2.x but fail miserably when trying to use them, through NAT, with a public Asterisk server. We have a bunch of Polycoms and Aastra phones that work great with NAT so I'm very familiar with NAT and Asterisk but not so much f...

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works great....

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/piperm...

...) * 7927 FETCH (FLAGS (\Deleted \Seen \Recent)) * OK Still here * OK Still here * OK Still here * OK Still here * 7940 EXISTS * 22 RECENT * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * OK Still here * 7941 EXISTS [At some point after this, something happens (or doesn't happen) and dovecot no longer notifies of new messages. The other client is still reading messages in the same mailbox and new mail is still arriving.] * OK Still here * OK Still here * OK Still here * OK Still here * OK Still h...

Full_Name: Axel Rasche Version: 2.1.0 OS: Win2000 Submission from: (NULL) (141.14.21.81) Hi, write.table does not accept the second statement with <col.names = NA, row.names = FALSE>. I do not see why this should not be possible. test = matrix(1:4, 2, 2, dimnames = list( c("a","b"), c("c","d") )) write.table(test, file = "test.txt",

I am having a problem registering my cisco phones which is exactly like that described in http://lists.digium.com/pipermail/asterisk-users/2011-May/262306.html except that I am on Asterisk 1.8.3.3 and using sip level POS3-07-4-00 The symptoms are: o 7960 lines show [X] o Outbound calls can be made from the phone, including call pickup of inbound calls, but not to it. o Trace shows REGISTER

...in R? I have to run this on a large dataset -- for a small set it is not a problem. ---------------------- Begin R code ------------------------------------ N <- 1005 * 14; myNorm <- data.frame(PaydexNormingCategory = numeric(N), SIC = numeric(N), CatMedian = numeric(N)); k=1; #j = 7941; ## For testing only for (j in levels(SIC)){ for (i in levels(PaydexNormingCategory)){ myData <- dfpaydex[(Paydex==i) & (SIC==j),]; myMedian <- with(myData, levels(Paydex)[median(rep(as.numeric(Paydex), Counts))]); myNorm[k] <-c( as.numeric(i), as.numeric(j), as.numeric(myMedian)...

...le, :page => ''synopsis'' %></p> <% end %> however if I do: <%= debug(@artist) %> from the same template, then my array of films never appears. Why is that, and is there a way I can make it appear? thanks dorian -- I do things for love or money -- +44 (0)7941 219 501 -- aim:oulalipo | yahoo:tachekent

Hi, I have a Cisco 7941 phone running SIP, and for a variety of reasons [1] have configured this in the short term to use my local Asterisk server to register to, and then have my local Asterisk server in turn register to the upstream SIP proxy at my voice provider. Voicemail is supplied by my VoIP provider, so I've...

...I'm off on the wrong track ;-) Thanks, Reuben -------- Original Message -------- Subject: External voicemail and MWI on internal phone Date: Mon, 17 Apr 2006 10:35:07 +1200 From: Reuben Farrelly <reuben-asterisk-users@reub.net> To: asterisk-users@lists.digium.com Hi, I have a Cisco 7941 phone running SIP, and for a variety of reasons [1] have configured this in the short term to use my local Asterisk server to register to, and then have my local Asterisk server in turn register to the upstream SIP proxy at my voice provider. Voicemail is supplied by my VoIP provider, so I've...

Does anyone have a solution for remotely getting the newer cisco phones (7941, 7961, 7970, etc ) to reread their configs (or even rebooting). I am running SIP firmware connected to asterisk. Check-sync doesn't seem to work anymore, I can't login to the phones as root because I am given a "challenge: <random digits>password:" prompt. occasionally I n...

Hi, (I think) I followed instructions here ( http://www.voip-info.org/wiki/view/Firmware+issues+on+7940+-+7960 section "Notes added Nov 2005, revised May 2006:" at the bottom of the page) to factory reset a Cisco 7942 I wanted to configure to SIP firmware. When booting, I can see this requesting and obtaining file term42.default.loads from TFTP server. Then it would send a request

...9;':include_blank => true'' in a date_select form, the result is always applied to the date field in the db as eg. Jan 01, which means that as far as I can see blank values aren''t possible. Am I missing something? thanks dorian -- I do things for love or money -- +44 (0)7941 219 501 -- aim:oulalipo | yahoo:tachekent

...be the best way to do what I initially though should be done like this: self.running_time.strftime("%M:%S") ? It''s taking me a while to get the idioms and the heuristics for RoR... Thanks in advance for any help available! dorian -- I do things for love or money -- +44 (0)7941 219 501 -- aim:oulalipo | yahoo:tachekent