search for: alows

I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would expect the RTP session should ideally be between the two end points of the call, in my case the AS5300 and the 7940 which are connected on the same VLAN as the Asterisk server. When I sniff the packets on the VLAN I find that all RTP packets are being relayed by the Asterisk server causing increased load on the

If there is a separate sieve mailing list let me know. But have question if someone else has done this or not. Right now I have web form people use to setup their vacation replies using Sieve (1.1.6). The form alows them to set what they want their reply address to be, whether to send the reply once, weekly or bi-weekly. I would like to setup another option to let them set the date in which to de-active the vacation replies automatically. Right now they have to remember to log back into the form to turn...

Full_Name: L. Petr?k Version: 2.2 OS: w2k Submission from: (NULL) (193.109.177.11) # rcmd build Translation-it * checking for file 'Translation-it/DESCRIPTION' ... OK * preparing 'Translation-it': * checking DESCRIPTION meta-information ... ERROR Malformed package name See the information on DESCRIPTION files in section 'Creating R packages' of the 'Writing R

Has anyone any experiance with the FileColumn plugin. I''m trying to create thumbnails, and also alow non picture uploads. Here''s some working code, but if I try and upload a zip, it says ''File invalid image''. Is there anyway of having both image uploads (and thumbnail generation) while also allowing non image uploads? class Entry < ActiveRecord::Base

Hello, Based on replies (which I'm gratefull for) I guess the only solution is to create a package, that will contain function to build lm()-like linear models, and allow weighting cases (as I described two days ago). This would also alow buliding those models for data in aggregated form without a need for using e.g. expand$grid(). Thank you again for replies.

Would anybody have any numbers on how large a box would be required to convert 100 or 200 SIP calls to IAX2, without transcoding, echo cancel, .. Or a setup with individual IAX2 calls coming on one side, and trunking being used to 1 or more remote boxes on the other side, to improve bandwidth usage ? It doesn't matter if you don't have a test done for exactly 100 or 200 calls, I'm

Hi, Im trying to register to a SIP provider that told me that they only need to authenticate using IP. the following string generates response 405 register => asteriskIPaddress@SIPproviderIP:5060 doing the following is not alowed by asterisk register => @SIPproviderIP:5060 any ideas? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/

I was wondering if there is a web front end for clients. By this I mean, is there something I can install on my apache server( wich is running on the same server ) that will alow my users to go to a page, login, and it drop them into there home directory. I have researched WEBDAV but I don't think it has the functionality that I need. Any help is greatly appreciated Wayne Dozier Network

Hi OpenSSH'lers! While using OpenSSH for quite a while, I became annoyed with the inflexible config-file parsing algorithm. I special it did not alow me to express: "Use *no* proxy for host xyz, but *this* proxy for all other hosts". So I had a look at the source an make a quick-n-dirty change, allowing me to use the special ProxyCommand "None" to express "don't

sorry to come with such a silly request but i have a code: shorewall-zone { "$name": SHTYPE => ''ipv4'', PARENT => "$PARENT" } and it bring a syntax error : debug: importing

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Hi, Sorry for my bad english but I?ll try to explain my problem I got an Asterisk running in my house with ADSL... I?m using X100P and TDM400P cards.... My intention is get calls via PSTN to my house and Redirect to my computer in my work using X-Lite by SIP... Here?s the map with Firewalls Call for anyone to my house => PSTN => X100P => EXTENSIONS => SIP/RTP => ISA MICROSOFT

Here''s a simple one... When using in_place_edit fields, they don''t appear unless they already have a value. How can I get similar functionality when they are empty? Thanks, Nathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://wrath.rubyonrails.org/pipermail/rails/attachments/20060511/a98640cf/attachment.html

...5 and Windows NT versions of the thing (with maybe a Mac or two thrown in for good measure...). The problem is that 1) I need to read strings, and 2) those strings may have various encodings. In the same sheet of the same file, some cells may be latin1, some UTF-8 and some CP437 (!). read.xls() alows me to read those things in sets of dataframes. my problem is to convert the encodings to UTF8 without cloberring those who are already (looking like) UTF8. I came to the following solution : foo<-function(d, from="latin1",to="UTF-8"){ # Semi-smart conversion of a datafr...

Is any way to detect changed MAC adresses? Someone taught change MACs peoples in my network and I have problems. E.g. Two computers working on one MAC, and one IP (static ARP and DHCP). WinXP is screaming some message... that two computers or more have the same IP. How can I find out who''s changed MAC?

I just read some of the discussion on the list about 'bitrate peeling' and remembered an interview of Monty that I have read recently. In it he says that Vorbis uses MCDTs <sp> and that these are theoretically reversable. And now, I learn that theoretically we can use bitrate peeling to make smaller files from larger ones, and that leads to my question. Could I theoretically

No, just no. You do not use goto to skip a code block. You do not return an obvious variable from a singly-inlined function and give the function a return value. You don't put unexplained comments about kmalloc in code which doesn't do dynamic allocation. And you don't leave stray warnings around for no good reason. Also, when possible, it is better to use block scoped variables

No, just no. You do not use goto to skip a code block. You do not return an obvious variable from a singly-inlined function and give the function a return value. You don't put unexplained comments about kmalloc in code which doesn't do dynamic allocation. And you don't leave stray warnings around for no good reason. Also, when possible, it is better to use block scoped variables

...y > > bringing up this topic at LSFMM last two years... > > You're right - but you have chosen the uneasy path. Yes. > Fixing __vmalloc code > is easy and it doesn't require cooperation with maintainers. But it is a hack against the intention of the scope api. It also alows maintainers to not care about their broken code. > > > He refuses 15-line patch to fix GFP_NOIO bug because he believes that in 4 > > > years, the kernel will be refactored and GFP_NOIO will be eliminated. Why > > > does he have veto over this part of the code? I'd...