similar to: Start asterisk on boot

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

Hi Our sip provider has two servers that sends calls to our asterisk 1.6. When server 1 sends call everything is working, but when server 2 sends call I get [Jan 13 14:56:23] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303390' rejected because extension not found. And the provider get an "404 not found" error on their side. What

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

Hi When i try to transfer calls from my ip2002 phone in asterisk 1.6, I can make the transfer and it rings on the extension I transfer to, but when I accept the call, asterisk dumps. How can I get it to work? And how do I save the dump error? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

Hi I have asterisk 1.6 and running queues with realtime mysql. I am trying to set another musiconhold then "default" but I cant get it to work, I have an musiconhold entry in my queue_table, but don't know what to put in there and where to put the file. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm,

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

Hi I have this in my queue extension and I see this in asterisk when I call to the queue, but no file is created in the directory any ideas? exten => s,1,Set(MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-${UNIQUEID}) -- Executing [s at kundservice:1] Set("SIP/0850001175-b7942770", "MONITOR_FILENAME=/var/spool/asterisk/queuecalls/QSAMPLE-1228382046.12") Regards

Hi Is there a user client that a group, like customer service can use? We have today an avaya IP-office with phonemanager pro and I want something equal to phonemanager pro, where you can logon to ques and see how many calls is in that queue and so on. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir:

We have 40 nortel ip2002 phones connected to asterisk 1.6, the problem I have is that the phone looses the connections with the server and then drops calls, we can reconnect but the customers don't like it. Anyone has the same problem? /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir: +46-(0)707548074, vxl:

Hi Im using a sip phone SPA921, and the one that calls me can hear me but I cant hear them, when I make the call I can hear them. Im running asterisk 1.6 behind a firewall, I have port 10000-20000 for rtp and 5060 for sip forward to my asterisk. Any tips? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Sveav?gen 56C, 111 34 Stockholm, Sweden Dir:

Hi [Feb 17 07:59:45] WARNING[21539]: channel.c:3477 ast_request: No channel type registered for 'USTM' [Feb 17 07:59:45] WARNING[21539]: app_dial.c:1502 dial_exec_full: Unable to create channel of type 'USTM' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) I get this after I restart my asterisk 1.6, it all worked yesterday. I have the

I've had good luck using the llvm-gcc & llvm-g++ on small projects, but I just discovered that it's apparently missing some of the GCC intrinsic functions -- specifically, when I try and compile VXL (http://vxl.sourceforge.net) it dies when it encounters __builtin_bswap32 . This is on OS X with the llvm-gcc-4.2 & llvm_g++-42 that are part of the XCode 3.2.3 I don't know if

Hi Kent, I suggest you open a bug report with a preprocessed testcase. Best wishes, Duncan. > I've had good luck using the llvm-gcc& llvm-g++ on small projects, > but I just discovered that it's apparently missing some of the GCC > intrinsic functions -- specifically, when I try and compile VXL > (http://vxl.sourceforge.net) it dies when it encounters >

Hi, Running Redhat 7.3 with SAMBA. I am trying to have a shared folder that everyone in the office can use an write to. I have one problem, as users open a file, it changes the user flag and only that user can open it in the future. All users are in the same group = Users Please help Regards Michael Crocombe Hi, If I can be of any further assistance, please don't hesitate to contact

Hi I have a couple of questions: 1. How do I exist from a domU console back to dom0 console (after issuing xm console [domU])? 2. How do I setup a dom0/domU in order to get a tty login? TIA Paolo _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi, I''m running Xen 4 on Slack64. I haven''t console login prompt to domU until I changed xvc0 to hvc0 on /etc/inittab file. No I have access to login but whenever I try to login I get this error from /var/adm/secure file root@darkstar:/var/adm# cat secure Apr 28 22:49:22 darkstar login[1286]: ILLEGAL ROOT LOGIN on `hvc0'' Apr 28 22:49:45 darkstar last message repeated

Hello All I just recently adopted a server with Gentoo/Xen as domain0 server with several domainUs attached to it. Please forgive my xen-noobness, but I was hoping that someone can help me out. These images have not been updated in a long time, and I''m trying to get all the domainUs updated with their appropriate services patches, etc. I''m currently working on one Gentoo

I decided to upgrade from openSuse 10.2 to openSuse 10.3. I installed xen things, but when I boot using xen option the system is blocked. It appears the green window, with little suse logo in the middle, and mouse cursor, but the system crashes. My computer is a Pentium 4 1,5 GHz, 512 MB of RAM. I attached "diff boot.msg boot.omsg" (as diff opensuse-normal-kernel opensuse-xen-kernel)