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Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

I am currently testing a 57i unit. No problems configuring the phone's config via phone/web UI. We are trying to avoid using the web UI, the reason is it will take a long time typing the softkey xml applications URIs on each phone, so we chose TFTP. Tried configuring the phone via a TFTP config server, but no changes took effect. I wonder why it doesn't work with TFTP even if I was able

I''m trying to get capistrano to deploy to an osx machine - rake remote:cold_deploy but it fails on the svn co line with "bash: line 2: svn: command not found" I''ve set the proper path to svn (/usr/local/bin) in both .bash_profile and .bash_login but capistrano seems to be ignoring these. How does capistrano know where to look for svn? Thanks

I'm stumped, can anyone find my error in this sequence. for(j in 1:(varsize[4]-1)) temp <- get.var.ncdf(nc=input, varid="p_foraging",c(1,1,j),c(varsize[1],varsize[2],1)) filled.contour(x, y, temp, color = terrain.colors, plot.title = title(main = paste("Everglades Wood Stork Foraging Potential \nYear", (2000+j)), xlab =

I thought I could try to post this question here since I have at times found excellent support here on A@H issues as well as pure/straight digium/asterisk issues. I am looking to finally build a permanent asterisk server and would prefer, if at all possible, to stick with the "new" A@H called trixbox. Does anyone know if the CentOS behind trixbox is a relatively complete CentOS system? I

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

This is more of a design issue I''m wrestling with, and was wondering what other people are doing. I have a products table with status values of In Stock, Out of Stork or Discontinued. This is unlikely to change any time soon. Approach A, is to have thses actual values in the status column, at least the meaning is very apparent, yet querying for products with a specific status could

Hi there, I've looked through the very helpful advice about adding fitted lines to plots in the r-help archive, and can't find a post where someone has offered a solution for my specific problem. I need to plot logistic regression fits from three differently-sized data subsets on a plot of the entire dataset. A description and code are below: I have an unbalanced dataset

We have set the do not disturb feature on our polycom phone such that incoming calls will not be rejected and sent to 'Busy' status. The user can still toggle dnd on/off, but incoming calls will still get in, indicated by the blinking light and the screen status. We were able to do that by setting call.rejectBusyOnDnd="0" in sip.cfg. However, the problem is that there is still

Hi there. Someone might want to notice that - compared to the official rsync man page - this page has a lot more information about each option, when to use and what to expect: http://www.ss64.com/bash/rsync_options.html Thanks Hiran Chaudhuri _____________________________________________________________________ Der WEB.DE SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen!

We're encountering no audio issues in ConfBridge. Only the moderator and the 1st invited have audio. When the 2nd invited number picks up the phone, only the announcement is heard, then followed by silence. This occurs frequently. Only 1 out of 5 test calls do not have this problem. We were hoping that switching from MeetMe to ConfBridge will solve the no audio issue, since there's no

...animal fine-scale movement in urban areas, or migration and range dynamics in the context of climate change. The primary mission of the Data Manager will be to manage the group's database system: we run a PostgreSQL/PostGIS database system on a Linux server, which hosts telemetry data on wood storks, sea turtles, raccoons, and associated GIS layers. In the near future, we will be adding advanced data from albatrosses, including accelerometers and infrasound. The Data Manager will manage the database server (database design, manual and automated data entry, metadata, set permissions for users a...

The MabLab at the University of Florida's Fort Lauderdale Research and Education Center (Davie, FL) is looking to fill an immediate opening for the position of a Biological Scientist. We are looking for an enthusiastic, highly motivated and responsible individual to join an exciting research team focused on spatial ecology, from fine-scale movement and habitat selection to range dynamics.

This patch removes support for use of NFS as a root filesystem from the kernel. It also updates the documentation to reflect this. b/Documentation/nfsroot.txt | 97 +++--- b/arch/arm/def-configs/a5k | 1 b/arch/arm/def-configs/adi_evb | 1 b/arch/arm/def-configs/adsbitsy | 1 b/arch/arm/def-configs/anakin | 1

This patch removes IP autoconfiguration (CONFIG_IP_PNP and friends) support from the kernel, and ensures that using NFS as the root filesystem (CONFIG_NFS_ROOT) will no longer work. b/arch/alpha/defconfig | 1 b/arch/arm/def-configs/a5k | 1 b/arch/arm/def-configs/adi_evb | 4 b/arch/arm/def-configs/adsbitsy | 4

I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running A@H 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on

I'd like to have a list of variables used in Asterisk 1.4, and which ones from v1.2 were deprecated/changed. Ex. Since switching from 1.2 to 1.4, nothing shows up when I want to display the value of ${TIMESTAMP}. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061123/6a11f804/attachment.htm

We recently installed Asterisk 1.4.2 Tried to make calls using the Originate command (Asterisk Manager Interface) All of the calls have zero billsec in the CDR. Stumbled upon this: http://bugs.digium.com/view.php?id=8680 so I guess the fix is not yet in 1.4.2. Is this fixed in 1.4.3/1.4.4? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Installed Asterisk 1.4.4 hoping that this bug (http://bugs.digium.com/view.php?id=8680)<http://bugs.digium.com/view.php?id=8680> is fixed. It was fixed, however a new problem came up. It seems that AgentMonitorOutgoing(c) is no longer able to change the 'channel' value in the CDR into Agent/<AgentNumber> format. This feature is very useful to us because we can identify calls