similar to: PROBLEMS with XP and SAMBA

Displaying 20 results from an estimated 10000 matches similar to: "PROBLEMS with XP and SAMBA"

2003 Sep 28
1
Forwarding SIP over IAX problem: No One Available
I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>
2010 Apr 23
2
How do you change library location ? (in R under windows XP)
Due to the new R 2.11 release, I want to implement Dirk's suggestion here<http://stackoverflow.com/questions/1401904/painless-way-to-install-a-new-version-of-r> . So for that I am asking - How can I (permanently) change R's library path? (The best solution would be one that can be run from within R) Thanks, Tal ----------------Contact
2014 Feb 19
3
Best way to virtualize Windows XP on Centos
I may have a need to run some version of Windows (XP?) on my desktop. As this will likely be a short-term thing, and since I have never used Windows, I would like to do this in the most painless way possible. A method that requires me to make the least changes to my Centos computer would be nice, since I'll probably want to back it out again later. I have never used any of the current
2005 Oct 15
7
You ASKED for an Asterisk book, you GOT an Asterisk book!
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons
2003 Dec 01
3
Re: Asterisk behind NAT << How to do it. (Leif Madsen)
> I'm pretty sure that is incorrect. The inside_net is the ip address of > the asterisk server, and the inside_mask is the subnet mask. At least > that is how I have mine setup in my sip.conf, and it works. > > inside_mask for the internal mask would make more sense to me as well :) > > -- > Leif Madsen <leif@hacklocalhost.com> > http://www.hacklocalhost.com
2008 Feb 19
3
simple usage of "for"
Hi list I have a data frame I would like to loop over. To begin with I would like crosstabulations using the first variabel in the data frame, which is called "meriter". > table(meriter[[1]], meriter[[3]]) ja nej Annan 0
2013 Nov 23
0
[LLVMdev] "Mapping High-Level Constructs to LLVM IR"
On 11/23/2013 12:18 AM, Mikael Lyngvig wrote: > Thanks, you have a lot of valid points there. I have myself long ago > abandoned the path of using C as a backend language due to the very > factors you mention. > > However, as I said, the document was put together in 30 minutes. Not > exactly ready for prime time :-) > > I do agree that all of the things you mention
2003 Sep 21
2
Incoming phone line rollover / hunt?
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I have a simple question about incoming phone line rollovers. How are these usually done? Is this done at the phone company usually, or is this something that Asterisk or channel bank is capable of? I just need someone to give me a brief explanation how it usually works, and if someone was implementing an Asterisk system, how they would go
2003 Sep 30
5
* not logging CDR to MySQL - anyway I can debug this?
Hi all, I think I've run out of options in terms of what I know about this. I have created a user called asteriskuser and granted all privileges to the asteriskcdrdb database. Then I created the table via the cdr_mysql.txt file. I have edited the cdr_mysql.conf file to reflect this, and added load => cdr_addon_mysql.so after compiling it from the latest CVS. If I check the
2003 Sep 11
1
How much to charge for Asterisk installations?
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a medium sized business that is interested in implementing * as their PBX system. They currently have a Panasonic system with Panasonic handsets that they are going to replace Asterisk with, as the current system is maxed out, and they don't even have voicemail capabilities. I have been considering using an Adtran Atlas 550 with FXO and
2003 Sep 06
7
OT: Creating documentation using a web interface
Hello all, I would like to document some things I am doing with asterisk, but would prefer to do this from a web interface. I am unfamiliar with any software that allows you to create online documentation from a web interface. Ideally I will be able to create documentation online from a browser, which then when saved, is immediately ready to be read online. Perhaps I can setup different authors
2003 Jul 16
2
list to data frame
Dear R helpers I am trying to convert a list into a data frame but when I try, I get a stack overflow error (Error: protect(): stack overflow). My list contains about 17000 rows and looks like shown at the bottom. The reason that I want to convert it in to a data frame is that I want to export it to a mysql database with the dbWriteTable function. The function that I use is
2004 Aug 08
2
asterisk-update script
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Here's a version I modified which grabs either a development or stable verision, and does a backup before updating from CVS. It also asks for addon's and cc. Leif Madsen did the original development and Mark released it. My changes does the minimum changes to previous version, to get what I need. It does the same version checking as
2003 Jul 10
1
Voicemail answers, but drops SIP call after about 3 seconds.
I am calling from my laptop to an asterisk box which answers the call and I can hear the voicemail prompts, but the problem is that after so many seconds, MSN Messenger drops the call because it thinks it hasn't been answered by the remote machine. I'm not sure if this is an asterisk problem, or if it is Messenger not knowing the call was answered. Has anyone else run into this sort of
2003 Aug 13
1
I can't get a two way conversation going?
I have tried both G711u and GSM codecs, and I get the same problem with both. The asterisk computer is running a TD20B card with two phones attached. I call from my laptop with a microphone to the asterisk box. Phone rings, I answer and the call doesn't drop. I can talk into the phone and hear myself on the laptop, but I am unable to get the sound coming into the laptop on the microphone to
2003 Sep 15
1
Radio for Music on Hold?
I'm curious if anyone has used a radio for MOH? If so, how did you set it up? I have a client who is interested in using a radio for the music on hold, since that is what they did with their old phone system. Thanks, Leif Madsen.
2003 Sep 17
1
Prices for new channel banks, patch panels, cables etc.. etc..
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi All, I'm having a tough time trying to find prices from dealers in Canada for some equipment. I am trying to implement an Asterisk box into a small business using 24 FXS ports and 8 FXO ports. I need to find the pricing for all the relevent equipment: cables, patch panels, channel bank chassis, cards etc..etc.. I think I'm going to tie
2003 Sep 29
1
Needed: Configuration Examples for VoIP Providers Asterisk can Register With
Hi all, I would like people to email me at 'leif at hacklocalhost dot com' some example configuration files for VoIP providers which * can register with. I am going to expand upon the FWD php "wizard" I created for these other providers, but I need some examples as I don't actually use anything but IAXtel and FWD. So far sipphone and iaxtel has been mentioned. I can
2003 Oct 06
1
MP3s in /var/lib/asterisk/mohmp3 causes Asterisk crash
Hi All, I have just compiled the newest version of mpg123 on a RedHat 9.0 system (mpg321 has not been installed) and I am using the newest CVS version of asterisk. Whenever I place any mp3 files in the /var/lib/asterisk/mohmp3/ directory, Asterisk crashes a horrible fiery death. If mp3s exist in that directory, then I can't even start Asterisk. If I start it without files then copy
2004 May 22
2
loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
Afternoon all, I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing