similar to: No Mailbox Subscription in SIP Users Suddenly

Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box

Hi all, I''m trying to get a pci card working on a guest os (ast01) on dum0 I hided the pci card w/: # modprobe pciback hide=''(0b:08.0)'' then added this line to the guest os config file (/etc/xen/ast01) pci = [ "0000:08:0e.0" ] Whenever I try to start it fails w/: # xm create ast01 Using config file "/etc/xen/ast01". Error: function takes exactly 4

Hello, pvshankar wrote > > Hello all, > > I have a data frame with column names s1, s2, s3....s11 > > I have a function that gets two parameters, one is used as a subscript for > the column names and another is used as an index into the chosen column. > > For example: > > my_func <- function(subscr, index) > { > if (subscr == 1) > { >

Hello all, I have a data frame with column names s1, s2, s3....s11 I have a function that gets two parameters, one is used as a subscript for the column names and another is used as an index into the chosen column. For example: my_func <- function(subscr, index) { if (subscr == 1) { df$s1[index] <- some value } } The problem is, I do not want to create a bunch of

Hi Jayson, 2008/1/11, Jayson Anderson <jayson.anderson at gmail.com>: > Greetings Arnaud, > > I'm sorry for taking up your time, but you are the primary POC for the NUT > project: > > I was just wondering if you know of any planned support for the Belkin > F6H550-UPS within the NUT project ? > > I am not a coder and as such, completely at the mercy of the

We have a great many VMs that are stuck on Lenny because of this. Probably can build it ourselves, but would prefer to avoid the maintenance nightmare. Please make this happen, as we really like Debian. Thanks, -- Jayson Vantuyl jayson at gogrid.com<mailto:jayson at gogrid.com> 417-207-6962<tel:+14172076962> (mobile) ________________________________ The information contained in

Hi Jayson, I forward your request to the right place. Carlos (or some other) will have an answer, or at least some questions for you. -- Arnaud 2008/6/27 Jayson Anderson <jayson.anderson at gmail.com>: > Greetings Arnaud, > > I wrote you some time ago about whether NUT would support the Belkin > F6H550-UPS device in the future. > > You had replied and mentioned to send

I have troubles installing the "SparseM" package on my Debian stable Linux system. Debian's version of R is: platform i386-pc-linux-gnu arch i386 os linux-gnu system i386, linux-gnu status major 1 minor 5.1 year 2002 month 06 day 17 language R This is the installation output: > R CMD INSTALL -l /usr/lib/R/ SparseM_0.36.tar.gz * Installing

Well, this is weird. After receiving a sip subscribe message from peer 2529266, here's what Asterisk responds with: -- (14 headers 0 lines)--- Found user '2529266' Looking for 2943110 in bell_CallStart (domain ua2.ipt.xxx.com) Dec 26 10:19:34 NOTICE[27345]: pbx.c:1741 pbx_extension_helper: Cannot find extension context 'bell_CallStart' Transmitting (no NAT) to

I am having a problem with asterisk voice mail messages that seems to be intermittent. Though the problem occurs most of the time, on rare occasions it will work fine - rare enough that I can't pin down what it is that works. The problem is that voice mail message get played back garbled. Occasionally, I can make out moments of a voice or another sound that may be in the actual message,

Hi, I have a problem that the machine running asterisk 1.6.2.11 freezes unexpectly time to time. Sometimes it runs for 4 weeks without any problem, sometimes after a free it freezes again in 24 hours. But usually it runs normally for 1 month or so before it freezes again. I could not find any additional info in any file located in /var/log/ including asterisk's messages file. Verbosity

How do you make a button inactive, so that it is visible, but it will not respond to clicks? Thanks Jayson

Can anyone tell me what I need to do to tell the voicemail and voicemail2 applications to download to a different slot? The only problem I have remaining is this one... I dial the voicemailmain extension from a PT480, and the display tells me "services is full, download refused", but asterisk PBX only occupies the first slot (I changed this)... Any help would be greatly appreciated.

Good Afternoon, I have an Asterisk box that is acting like it is passing "core show" before every command I type. For example, if I type sip, I will get "No such command 'sip' (type 'core show help sip' for other possible commands). Any ideas? -- -jayson

Hello Everyone, We have an order for SMS messaging. Can you gents and ladies be kind enough to disclose if SMS is possible using Asterisk? What is a quick way to test a `Hello World` to my cell. Finally, do all service providers support SMS messaging? Kind Regards, Jayson. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Using realtime for SIP. Using standard DB schema. Tried mailbox as varchar(50) and bigint(10) "sip show peer XXX" shows "Mailbox: " (empty) So MWI isn't working This happened before when we tried to go up to 1.8, so we stayed at 1.4 We're forced to go to 13 now. Obviously a lack of MWI is a big issue for 500 units. Any idea why we're not getting it? We're

Dear R People: I am attempting to install SparseM on R 2.15.0 on a Linux 11.10 system. Here is the output > install.packages("SparseM",depen=TRUE) Installing package(s) into ?/home/erin/R/x86_64-pc-linux-gnu-library/2.15? (as ?lib? is unspecified) --- Please select a CRAN mirror for use in this session --- Loading Tcl/Tk interface ... done trying URL

I''m currently trying to build modules against the kernels created with Xen 3.0.5rc4. This used to not be such a problem, as Xen created a kernel directory and the built in it. Plain Jane, nothing fancy. I''ve noticed that somewhere since I did this (which was as recent as 3.0.4-1) the kernel build now does things a bit different. Apparently there is some sort of

Hello Everyone, We are looking to transition our 23 channels from testing/lab into production. During testing we used the free open source g729 license using the instructions found here: http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/ A little more about our setup. All recordings have been converted to G729, all voicemail messages are also in G729, finally

I've been asked to create a graphical "call-queue" manager. That is, use the existing call queues application but allow a way to view what's coming and attach information to it. As far as the "attaching information" that's in the realm of my application, but I'm trying to figure out if the internals of queues are exposed through any interface. Any help there?