similar to: Sip over VPN not working

Hi guys, I have a strange problem. I can authenticate a user with wbinfo from my domain controller (security =3D ads), however when I try and map a share, the authentication fails. i.e. # wbinfo -a 'COAL+bcanglo%bcpass' plaintext password authentication succeeded challenge/response password authentication succeeded # smbclient '\\xxxxxxx\timtest' -U 'COAL\bcanglo'

Hi guys, I have a strange problem. I can authenticate a user with wbinfo from my domain controller (security = ads), however when I try and map a share, the authentication fails. i.e. # wbinfo -a 'COAL+bcanglo%bcpass' plaintext password authentication succeeded challenge/response password authentication succeeded # smbclient '\\xxxxxxx\timtest' -U 'COAL\bcanglo'

SoftEther VPN Once setup, it just works.... Regards, Richard --- Richard Zimmerman Systems / Network Administrator River Bend Hose Specialty, Inc. 1111 S Main Street South Bend, IN 46601-3337 (574) 233-1133 (574) 280-7284 Fax -----Original Message----- From: centos-bounces at centos.org [mailto:centos-bounces at centos.org] On Behalf Of david Sent: Monday, April 04, 2016 1:57 PM To: CentOS

And openvpn. Avoid ipsec as it's too complex and pptp is unsecure. Eero 4.4.2016 9.55 ip. "Richard Zimmerman" <rzimmerman at riverbendhose.com> kirjoitti: > SoftEther VPN > > Once setup, it just works.... > > Regards, > > Richard > > > --- > Richard Zimmerman > Systems / Network Administrator > River Bend Hose Specialty, Inc. > 1111 S

Hi, I tried to turn on Video and get the following cli-WARNING output -- Executing [8600 at outgoing-kamailio:1] Answer("PJSIP/7000-00000000", "") in new stack > 0x7f46f41ff2e0 -- Probation passed - setting RTP source address to 192.168.8.203:17200 -- Executing [8600 at outgoing-kamailio:2] ConfBridge("PJSIP/7000-00000000", "8600") in new stack --

I'm new to Asterisk and have installed it on OS X. I am trying to use an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and am not having any luck. I have looked at several settings pages at voip-info.com but have had no luck. I am on a school network which I assume has some security but the particulars are unknown as they are not very helpful. It is necessary to

Folks, I have made an inbound call. Now I want to make an out bound. I have a sip softphone hitting the pbx. On my motorola firewall I have allowed all traffic in and out to my address. There is not a firewall on the PBX yet. here is the log off the softphone..and the PBX logs look like this Registration from 'Tommy Denton <sip:201@209.101.93.30>' failed for '

Hello, they are successful to start asterisk, task that the error that I had previously had had to a configuration problem. Start asterisk in modality consol and when two softphone speaks is not felt well, and I have the following error: -- Registered SIP '1000' at 10.0.0.7 port 5060 expires 1800 -- Saved useragent "X-Lite release 1103m" for peer 1000 --

I have d/l the iso (*@home 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to

Hello all: I'm having a strange problem with GNBD and LVM on two fully updated CentOS 5.5 x86_64 systems. On node1, I have exported a gnbd volume. lvcreate -L 500M -n mirrortest_lv01 mirrorvg gnbd_serv gnbd_export -d /dev/mirrorvg/mirrortest_lv01 -e node1_lv01 On node2 I have imported the volume: gnbd_import -i node1 Next, on node2 I attempt to create a mirrored LV with the

I have * box on machine with external ip address and internal one I'm tring to register to it from to machines - one from innternet ( everything is ok - in sip.conf nat=yes)\ and the other one is in the internal network (in sip.conf - nat=no ) and it say 403 Forbidden? Any Ideas ? here are the logs and configs From the external SIP Client whic registers.

Hi, I have a question about the "sext..to" instruction. In the document, I found two examples: %x = sext i8 -1 to i16 It means: i8 -1 = 1111 1111 --> 1111 1111 1111 1111 = i16 how can it determinate, that the i16 value %x positive is (65535)? And the second example: %y = sext i1 true to i32 1 --> 1111 1111 1111 1111 1111 1111 1111 1111 In this example, %y is -1 I'm not sure

On Mon, Oct 6, 2008 at 9:30 AM, Le Anh Quang <anh_quang.le at mailbox.tu-dresden.de> wrote: > Hi, > I have a question about the "sext..to" instruction. In the document, I found > two examples: > %x = sext i8 -1 to i16 > It means: > i8 -1 = 1111 1111 --> 1111 1111 1111 1111 = i16 > how can it determinate, that the i16 value %x positive is (65535)? > And

Hi, I'm analyzing freepbx extensions. When creating ivr with freepbx, it writes like this: exten => 1111,1,Answer exten => 1111,n,GotoIf($["${CONTEXT}"="from-internal"]?USERCID:SETCID) exten => 1111,n(USERCID),Macro(user-callerid,) exten => 1111,n(SETCID),Set(CALLERID(name)=${CALLERIDNAME}) exten =>

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Folks I would like to have my windows 7 laptop communicate with my home server via a VPN, in such a way that it appears to be "inside" my home network. It should not only let me appear to be at home for any external query, but also let me access my computers inside my home. I already have this working using M$'s PPTP using my home Centos 6 gateway/router as the PoPToP server.

hey all, I'm using openssh-3.5p1, was trying to set up a 'reverse telnet' session (sun solaris 2.6 on both machines). Anyways, I was doing: server% ssh -R 1111:<server>:2222 <client> client% ssh -p 1111 <client> where <server> is behind a firewall and <client> cannot reach <server> Anyways the idea was to connect to the socket on

I am using the latest xlite phone to connect to the latest version of asterisk (20040727). When I try to make a call the xlite phone tells me "Call not approved". I used the configuration options that were listed on the wiki. The context in the sip.conf file is "from-sip". I have a matching context listed in the extensions.conf file. The phone is able to register

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

I just installed Xorcom Rapid and I'm trying to connect with Xlite. In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address of the new install. I can ping that box. When I try to connect I get hung on the "Awaiting Proxy login information" and the log reads: ======================================================================== ? 2004 Xten Networks, Inc. All