similar to: Shared folder / Samba issue

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, search in web for regular expressions i get the information that the line below replace all AUTO string's like AUTOBAHN ,AUTORENNEN with 1 but nothing happend. Using the [] in the pattern it works like i'm expected, but i didn't want single character replacment. Where is my mistake? bcode <- gsub("/^AUTO.*/","1",MyStringVector,ignore.case=T,extended=T)

Hi We runs asterisk 1.6 on a ubuntu 8.04 server. How can I get asterisk to start at boot? I have created an file named asterisk in /etc/event.d and put in this # This service maintains Asterisk from the point the system is # started until it is shut down again. description "Asterisk daemon" start on runlevel-2 stop on shutdown respawn exec

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 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 I'm getting this error with my attempt at setting up dovecot for virtual mail accounts: Failed to create storage with data: /home/vhostpath/usersname/Maildir/ I cant establish any cause. I dont think its permissions as it was saying 'Permission denied' till I fixed that and the UID and GID are those of the directories owner. I was prompted bythe log files to set the minimum and

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, I''ve got a newbie question--sorry if this is covered elsewhere, I parsed through the archives for awhile and didn''t see it. I''d like to listen for whenever a file is renamed (e.g. foo.txt -> foo.old) and then magically change it back. This sounds odd, but I''m working with a stubborn application and this will actually make things work nice. So, if I do:

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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 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 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,

Hey all, This example of working code: @a = @b.where(:a_group_id.ne => nil).collect { |b| b.a_group } doesn''t make too much sense to me. For one, I am not sure what that "ne" is doing there. Is this a postgresql thing? Second, it looks like we are collecting active record objects where the a_group_id is null, collecting them in an array and then invoking the a_group

Hi folks, I've got trouble doing an blind transfer from an "EXEC Queue quename|t" in an AGI script. Attended is working fine, also when doing the same queue from the extension.conf file is fine. Here's my log; -- Executing AGI("IAX2/utv01-5", "agi://localhost/queuecall.agi?queue=vxl") in new stack -- AGI Script Executing Application: (QUEUE)

Hi, I'm trying to build 5-STABLE, I have cvsuped the latest source, cleared out /usr/obj and I still get this problem. Any idea what could be causing it? Mark ===> lib/libmagic cat /usr/src/lib/libmagic/../../contrib/file/Header /usr/src/lib/libmagic/../../contrib/file/Localstuff /usr/src/lib/libmagic/../../contrib/file/Magdir/zyxel /usr/src/lib/libmagic/../../contrib/file/Magdir/xdelta

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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 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: