similar to: one way rtp stream (Sent alwax to 127.0.0.1)

On Wed, 30 May 2018 15:26:37 +0200 "Stefan G. Weichinger via samba" <samba at lists.samba.org> wrote: > Am 2018-05-30 um 15:01 schrieb Rowland Penny via samba: > > > There are three main winbind backends, but only two are really used > > on Unix domain members, the 'ad' and the 'rid' backends. Which you > > use is really down to a simple

Am 2018-05-30 um 15:56 schrieb Rowland Penny via samba: > If the last two lines are actually in your smb.conf on disk and you > want use the 'rid' backend, then set the something like this will work: > > idmap config DOMAIN : range =310000-40000 > idmap config DOMAIN : backend = rid > >> >> the two lines >> >> idmap uid = >> idmap

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

Jerry, The 3.2.0 release is reporting this in the log.nmbd [root@beta samba]# cat log.nmbd [2008/07/02 12:26:52,? 0] nmbd/nmbd.c:main(849) ? nmbd version 3.2.0 started. ? Copyright Andrew Tridgell and the Samba Team 1992-2008 [2008/07/02 12:26:52,? 0] lib/util_str.c:safe_strcpy_fn(709) ERROR: string overflow by 1 (16 - 15) in safe_strcpy [192.168.100.20/24] Other than this issue; no real

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general at ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162

Hello List, following this thread: http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains I tried to configure on the pjsip.conf the same endpoint with different domains like: [1000 at sip.domain.com] type=endpoint [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console:

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

All, I've set up tinc yesterday (Win32 -> Linux), the connection comes up but I can't reach a host behind the tinc "Server". Maybe I did something completely wrong, so let me first describe my network.... The "Server" has multiple Interfaces, be we just care for two of them here. One NIC "eth2" ist to the LAN (192.168.100.x/24) and the oher one is

I have a linux sever running it is version 2.6.20-2936.fc7xen. The samba version is Version 3.0.27-0.fc7. My problem is XP and Windows 2k Machines connect very slow to the mapped drives. It does not matter if they are connected as the IP (\\192.168.100.32\*) or a mapped drive in the Windows Host file (Sambasales\*). Has anyone else had these problems, and if so what did you do to fix

Hai Stefan, Yes, its always better to ask the list, that way everybody can learn from it. ;-) > Do you think I will have to rejoin it to the domain? No i dont think so. Please note, o dont know anything about gentoo except that they have a good wiki/info pages. If this was debian, then in this case, what i would extra do here, run : samba -b and backup all folders of samba and any thing

Good day to all-- First off let me say that I have been very pleased with the mailing list. I have learned a ton of stuff just reading other peoples questions and comments. I really enjoyed the VOIP Conference call on Friday morning. Still working on figuring out the best approach to custom voicemail emails (the reason I joined this group); however, we have more pressing issues. I

Hi, I've seen some hardphones or Softswitchs now support this sip.instance feature : http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt I don't really see any convincing use of this draft but I would be curious to share thoughts on it. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL:

I''ve posted a lengthy description of an authorization plugin for Rails on my blog: http://www.billkatz.com/authorization It describes a proposed DSL for authorization, a pattern for use that describes conventions, and a reference implementation that lets you test the some of the ideas. I hope that some subset of Rails programmers gravitate toward a common DSL for authorization, which

Hello, I've created a routed network that forwards to a physical interface: <network> <name>default</name> <forward dev='eth0' mode='route'/> <mac address='52:54:00:f2:5b:4f'/> <ip address='192.168.100.1' netmask='255.255.255.0'> <dhcp> <range start='192.168.100.10'

Dear list, I was trying to use the "notify" action with dovecot sieve but I just couldn't get it work. The relevant lines look as follows: ,-------- |require ["fileinto", "notyfy"]; | if header :contains ["to", "cc"] ["userA at example.com", "userB at example.com"] { | notify | :method "mailto" |

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Has anyone else run into a problem with trying to use mock_model in spec/lib ? For some reason, I can take the same spec, put it in spec/models, have it run fine, but put it in spec/lib, and have it complain about not being able to find #mock_model Thanks, Edward