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Hello Thank you very much Carlow :-) ! Regards/D_S

Hello I have made a brochure which has the dimension 630(mm) 297(mm) which is made for printing jobs. the dimension 630 is then folded so it becomes 3 pages (you fold it lika a Z.) Why I have to do like this is because I have an illustrative picture on page 2 with explaining text Now I have to...

I had a total blackout :-) I thought that the line tool hade the same preferences as the box tool i.e that the line had an inbound color... Thanks anyway! Regards/D_S

Hello I have a very annoying problem, if I'm using the 'line tool' and drawing a line current line wont get the color that the swatches has? I dont know whats wrong... /D_S

---- Wrong ----------- If I dial from SIP-phone to CCM-phone => I can hear in SIP-phone what is said in CCM-phone but I cant hear anything in my SIP-phone... ---------------------- I ment: If I dial from SIP-phone to CCM-phone => I can hear in SIP-phone what is said in CCM-phone but I cant hear...

Hello Thnaks but... They can find eachother... If I dial from CCM-phone to SIP-phone => I can speak/hear both directions... If I dial from SIP-phone to CCM-phone => I can hear in SIP-phone what is said in CCM-phone but I cant hear anything in my SIP-phone... Greatful for help.... Regards/D_S

Hello I have a funny problem.... Earlier I had an asterisk (A1) connected to a CCM and it worked very good. Later on I had to switch machine and installed a new asterisk A2 that is connected to the same CCM as mentioned earlier. The CCM is the one I had when I was using A1 the only thing I...

Thanks for your book tip, I have found some examples on the web how to generate pages (XML) but where should they be put in CCM ? Regards/D_S

Hello Earlier I have been using SIP-phones from Polycom. On some of their modells you can use their display as a web browser. 7960 seems to be a phone of good quality but can I use its display as a web browser for display needed information for current user? If the answer is yes, how do I do...

Found it :-) In SIP-trunk settings there are some interresting options: Redirecting Number Delivery

Hello Lets say I have 2 CCM phones (P1 & P2) P2 has call "forward when busy" and this number is used by an SIP trunk i.e if P2 is busy it should redirect to nr 23XXXX which a route pattern takes care of and send it via my SIP-Trunk to an Asterisk. Lets say now that P1 making a call to P2...

Hello In my CCM I have registrated 2 devices ( devA and devB) devA has a DN 5040 and devB has a DN 5041, if I make a call between them everything is working nicely... but if I try to make a call from asterisk to CCM (devA or devB) its not working let me explain... In asterisk I have made a...

Hello There was som problem with the Kernel (?) a friend of mine solved it, thanks anyway! /D_S

Hello What I can remember I'm running 4.1 I cant control for the moment becuase we are switching offices so no servers are available. /D_S

Thank you jerake75 its working! I deleted all settings and followed your advice and it worked :-) A funny thing that I have noticed is even if I delete, in this case a route pattern, it seems like it still is in use(?) I know that becuase I deleted one and then made a trace with Ethereal and...

Hello I have installed VMware on Linux (Red Hat Enterprise) without any problems but when I try to install windows 2000 prof. it says that it cant find any cdrom? In the settings window wheere you choose to "start virtual machine" it says that my cdrom is on /dev/cdrom/ Any ideas what can be...

Hello Thanks you all for your help I have followed your advices but no tranformation is made,.... the server is about to get thrown out of my window of the 5th floor :-) .... Do I have to have any route group or route list when using route pattern or route transformation. As I know, with my...

Thanks for your answer but it doesnt work. I have tested that before but I get this message "Your call can not be completed" The funny thing is I have done a trace on my Asterisk server that my SIP-trunk should direct calls to and if Im dialing the number 95080 and do as you (pndscm) described...

Hello Lets say a user is dialing 95080 I want to remove first digit (9) and add +461111 before 5080 so it matches my id in my Asterisk so the final number/id will be +4611115080. That number/id is then used when using my SIP-trunk. How can I achieve this? Regards /D_S

Hello I'm playing around with 4.1. I have seriously problem to get this to work as I want. What I want to do is making CCM-phones speak with Asterisk-phones. I have done this: 1) Added a Devicepool and added devices to this devicepool 2) The devices above belongs to a specific user 3) I have...