search for: partenaires

Hello, I am using the odfWeave packages; I draw a table using the function odfTable: partCols <- gsub(?\\.?, ? ?, names(partenaires)) odfTable(partenaires, useRowNames = FALSE, colnames=partCols) it's working as I have a table in my output file. I would like to know how to change the background color for the header cells. I assuming i have to use tableStyles, but I don't understand how. Does someone have an idea to...

Hello R users, I found the odfWeave package to create an odf document. It seems to be a very nice tool. So i tried to used it to create a report with multiple plot: I create an odt file with some code inside: I connect to a mysql database I get a list of projects foreach project I would like to make a plot (a map exactly) then in a R console I use the odfweave (inFile, outFile) function.

Hello, I've been looking into OpenSIPS to see if it's a worthwhile addition to our setup. We're currently running a cluster, using Heartbeat, between two servers. It works well but I'm interested in seeing if we can improve it. My manager heavily uses RPM's for installations rather than source, particularly using yum to update. I'm trying to actually install OpenSips via

Hello, I noticed that a call on hold is disconnected after 5 minutes, whatever the value of the "rtpholdtimeout" parameter in sip.conf. Tested from v1.8.10.0 to 1.8.32.3. The version 1.8.8.0 is not affected. I don't know between 1.8.8.0 and 1.8.10.0. Does anybody has a solution to increase the timeout of a call on hold ? Is this a bug or a new parameter ? Thanks. --

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I'm using a few GXP-2000 with firmware 1.0.2.13 and everything seems to be working fine. However, there are a couple of issues I'd like to know if are possible: 1) Even though the phone has 4 line appearances, if I am speaking on a line, the phone can no longer receive phone calls. I can manually select another line and make calls, but when Asterisk tries to send a call to it, I

Here is the report and the complete diagram. And sorry for email problem and incomplete email ! I have made new test. Eth0 and eth2 are bridged. I can ping NET from LAN I can ping every firewall''s interface from LAN I can ping eth1 from private LAN I can ping everything from firewall Bridging is activated in shorewall.conf >From LAN i can ping 192.168.11.253 but not 192.168.11.254

Hi, I have the following config: [7960] <--skinny--> [Cisco CCM] <--SIP_trunk--> [asterisk] <--SIP--> [X-lite] Is there a chance to avoid the RTP stream from passing through the Cisco CCM ? I would like to have all RTP handled by the *. This is just a testbed, for a larger project. What I want to achieve, is actually this: [Cisco Phone] <--skinny--> [Cisco CCM]

Hi all, Did anyone had success with Q.Sig on * 1.2, especially with Alcatel 4400 (which seems to only support Q.Sig) ? I am thinking about interconnecting 15 sites together with asterisk (probably using IAX or SIP). I have a very heterogeneous environement using both PRI and BRI, but my pilot will start with the Alcatel at the central site. Any help/example is most welcome. BR, - Patrick -

Sorry some email problem, i have change it for more reliable one. I have try this morning to netmasq 192.168.11.0 (eth1) to 192.168.1.0 (eth0), but it is a mistake. Yes thank you for answering so fast ! I have corrected it, here the new diagram and the new routing table. But it still doesn''t work. From the router i can access to 192.168.11.254 I have add the rules : DNAT loc