similar to: SIP Anonymous auth

Hi there, I recently posted about problems with winbind resolving names from ADS. We have an ADS Domain Controller serving the domain MED-DUS and an NT 4 PDC serving the domain MEDOIL. We want to use a samba server as memberserver in the MED-DUS domain. I joined the samba server to the MED-DUS domain. The domain join worked ok. All users (including MEDOIL) are visible. Next i started winbind.

Guten Morgen Geoff Shang, Am Freitag, 14. Januar 2005 um 00:40 schrieben Sie: GS> The only thing I can think of that might cause your reencode not to work is GS> that perhaps you have MP3 files with extensions with capital letters (i.e., GS> not "mp3", but "MP3" or "Mp3" etc). This maybe could cause the match to GS> fail. thank you for the answer. i

On Friday, 14 January 2005 at 09:26, Carsten Henkel wrote: > Guten Tag Carsten Henkel, > > Am Freitag, 14. Januar 2005 um 08:59 schrieben Sie: > > CH> what is the alternate to ezstream for streaming without a gui ? > > is ices0.4 a good choice to stream & reencode playlists ? I think so. :) I'd be surprised if it ran on windows though.

Dear all here is my data called "rglp" structure(list(vzorek = structure(1:17, .Label = c("179/1/1", "179/2/1", "180/1", "181/1", "182/1", "183/1", "184/1", "185/1", "186/1", "187/1", "188/1", "189/1", "190/1", "191/1", "192/1",

Guten Tag Frits Letteboer, Am Mittwoch, 2. Februar 2005 um 14:40 schrieben Sie: FL> Carsten Henkel wrote: >> hi together, >> >> i have >> <directory> >> <yp-url-timeout>15</yp-url-timeout> >> <yp-url>http://dir.xiph.org/cgi-bin/yp-cgi</yp-url> >> </directory> >> <directory>

Hi all, I have two sipgate accounts (numbers), if I have both accounts register only one will work for incoming calls (which is all i'm interested in). However if I disable either account the other account will work perfectly. Am I missing something obvious? Thanks in advance, Ray. Excerpts from sip.conf - [general] 8<---- SNIP! ---->8 Register => 1212121:aaaaaaaa at

Hi there, is there any howto how do i configure a asterisk/trixbox for mail2fax? The fax must be send over sipgate or other SIP peers. (i dont have any "normal" telephones connected). What i wanne do is somethink like this: Subject: +49691234567 Attache: *.pdf The attched pdf have to be send ;) -- Mit freundlichen Gr??en Daniel mailto:daniel at listmails.de

Guten Tag Geoff Shang, thank you for this information. whe have to chanles in two different ports. i want to mount port 6000 mountpount /dsl as fallbackmount for the server that runs on port 8000. whis this way, i can only mount when sthe stream is on the same port. have a lot of thanks Am Mittwoch, 2. Februar 2005 um 09:09 schrieben Sie: GS> Hi: GS> The way to do this is to define a

hi there, i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x network, behind my main computer, i cam make call, receive calls, all works fine, with all providers except sipgate.de, there i can receive call and make them, i can hear the other end but they can not hear me, this is only the case with sipgate.de i don#t know how to configure it and thought maybe someone can help

Hi. I'm running Kernel 2.4.19 and Samba 2.2.5. I'm in need to do a backup out of big SAPDB database (60 GB) onto a share on a Win2K-box (all my databases have to backup onto this w2K-box where they get saved on tape). This works fine as long as it comes to 2 GB. Than it stops with the message "file size exceeded". Obviously there is a file size limit somewhere in Samba. How

Dear all, I have an asterisk sip issue which I don't believe is unique. I use a registrar (sipgate.co.uk) where I have 3 different accounts. These accounts provide me with three seperate local phone numbers which allow me to allocate them to seperate users. By using just one of these accounts I can set asterisk up to send and receive calls no problem. However, when I start to introduce an

hi, i want to reencode our ezstream. but it dosent work. mdplay and lame are installed on the server. the config for ezstream is in the bottom of this mail. i dont understand the problem. have you any ieda ? have a lot of thanks my config: <ezstream> <url>http://localhost:8000/backup</url> <sourcepassword>*****</sourcepassword>

Easy one. I have win98 clients that login and connect to the shares fine. My problem is the user home directory, each user can see their folders and can even open them but they cannot save to their home directory. Each users home dir is setup as 0700 (-rwx------), as are the files and subdir's, which I believe is Read, Write, Execute for the owner of the file. What am I missing? Chris.

Hi, Anyone here using sipgate.de ? It worked for months, but for a couple of days now I'm unable to register with them. My account is ok, because I can login to the website. Asterisk keeps showing me: Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I

Hello i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615 at sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '"01179553708" <sip:SIP-ID at sipgate.co.uk>;tag=as30eb9dd1' --

Sorry, Forgot to say that I am looking for free software On Wed, 9 Mar 2005 16:31:10 +0100, Carsten Henkel <carsten@chatlabel.de> wrote: > Guten Tag Mohamed Eldesoky, > > Am Mittwoch, 9. M?rz 2005 um 16:19 schrieben Sie: > > ME> I want to do live streaming from the mic from windows machine. > ME> Which applications do you recommend ?? > > sam3

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi all, I'm trying to resolve a weird issue with SIP routing. I have a number of SIP trunks, from a selection of providers, all of which are registered in sip.conf: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=yes tcpbindaddr=0.0.0.0 transport=udp bindport=15060 srvlookup=yes allowsubscribe=yes

Hi, have some problem with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de

Hi, I have a few accounts with sipgate.co.uk to get some different DiD numbers. However, when an incoming call comes in, it seems to pick the wrong peer from sip.conf which sends the call into the wrong context and it fails because there is no extension in that context to match the register. Using the config's below, if I dial the DiD on account 2222222, it works fine - picks peer 2222222