similar to: Limit on number of SIP channels?

Hi Chris, Le 22/03/2016 22:07, Chris Hastie a écrit : > I'm trying to remove a DC from a Samba4 based AD network, but run into > an error that I can't fathom. Can anyone point me in the right direction? > > # samba-tool domain demote -Uadministrator which version of samba are you using? 4.4 or below? is the sogo3.ad.oak-wood.co.uk server still running ok or do you have

I'm trying to remove a DC from a Samba4 based AD network, but run into an error that I can't fathom. Can anyone point me in the right direction? # samba-tool domain demote -Uadministrator GENSEC backend 'gssapi_spnego' registered GENSEC backend 'gssapi_krb5' registered GENSEC backend 'gssapi_krb5_sasl' registered GENSEC backend 'spnego' registered GENSEC

On 16/02/16 16:29, Chris Hastie wrote: > On 16/02/16 16:01, Rowland penny wrote: >> Do you have the ldb-tools package installed on the DC ? if not can >> you install it, then run this command: >> >> ldbsearch -H /var/lib/samba/private/sam.ldb >> '(&(objectclass=user)(samaccountname=*))' | grep chris >> >> Can you post the results. >

On 16/02/16 07:47, Chris Hastie wrote: > Hi > > I'm experiencing some odd behaviour when trying to change passwords. > I have Samba 4.1.6-Ubuntu configured as an AD-DC on Ubuntu 14.04LTS. > When I change a password (either from a Win10 Pro client, or using > smbpasswd on the machine itself) it all reports that things have > worked. I can then login to Samba using the

Yeah the "variable stays there" because the channel is never up to be cleared. If you do something like exten => _X.,1,Wait(1) exten => _X.,2,Hangup You will see the same behavior. Can you confirm?? I am running CVS from about a week ago... Alex > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com >

But it does indicated that a variable is staying assigned that should not be, which could have other impact over time??? The behavior is very different for c call where there is a dialplan match for the dialed number, when the call completes the channel extension variable is cleared. If you do not mind please ad a bug note that you experienced the same thing! The bug marshals think I am nuts.

I se what you are talking about I an able to reproduce!!! However your PRI may be in a Round-Robin picking order, that would cycle through all of the channels until it reaches an end and then it repeats. I set our PRI to first available hunting instead of RR and it will use the same channel over and over again regardless if the call exists. If anything it's a feature!!! Unassigned DID will

Could some out there with a PRI check and see if this problem shows up on your system? The test is to dial a number routed to * via a PRI where there is no match in the dial plan for the dialed number. Asterisk will reject the call, but "show zap channels" still shows the channel assigned to the number that was dialed under the extensions column. The channel WILL answer another call,

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

I tried that, you have to ANSWER before you can clear it, which is not a good idea... > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Alexander Lopez > Sent: Tuesday, September 13, 2005 9:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users]

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

Hello All, I would like to record inbound and outbound calls to and from one number. I tried to add lines to my extensions.conf: DAY=`date "+%m-%d-%y_%H:%m"` ;outbound exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(${TRUNKL3}/${EXTEN}) ;Inbound [line2] exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(SIP/101,20) exten =>

On 16/02/16 16:01, Rowland penny wrote: > Do you have the ldb-tools package installed on the DC ? if not can you > install it, then run this command: > > ldbsearch -H /var/lib/samba/private/sam.ldb > '(&(objectclass=user)(samaccountname=*))' | grep chris > > Can you post the results. Here you go, without any changes to generic names (ie I've kept my

Hi, I'm monitoring an Asterisk 1.2.18 box because sometimes I get two Snom 360 phones creating a lot of SIP channels between them and it seems they never die. How can it be? Thank you. Giorgio A "show channels" excerpt follows: SIP/20-08a7aa80 (None) Up Bridged Call(SIP/31-08a64220) SIP/31-08a64220 263 at inbound:1 Up

Hi all, I have a Cisco 1760 gateway and and Cisco 7960 VoIP phone running via Asterisk. Both are running g711A codecs and SIP. On inbound calls I get a native bridge, however on outbound calls I never get a native bridge. With other SIP gateways I do get a native bridge on the outbound call. My sip.conf is as follows: [cisco1760] type=friend context=incoming host=192.168.0.55 insecure=yes nat=no

Christoph Lehman had problems with seperated data in two-class logistic regression. One useful little trick is to penalize the logistic regression using a quadratic penalty on the coefficients. I am sure there are functions in the R contributed libraries to do this; otherwise it is easy to achieve via IRLS using ridge regressions. Then even though the data are separated, the penalized

I have something going on that I don't fully understand after a weekend of looking for answers. I have an iax account with Tesco that works flawlessly with the Zoiper client - but is giving me trouble with inbound calls in Asterisk 1.6. After some playing I have ended up with an iax.conf file that looks like this: [general] calltokenoptional = 77.75.0.0/255.255.248.0 maxcallnumbers = 16382

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

On 16/02/2016 14:55, Rowland penny wrote: > This is strange, just logging in shouldn't create a user in AD and when > you see MYDOMAIN\chris this is just winbind i.e. > > How are you logging into the DC that causes the creation of a user in AD ? From another machine, in an Ubuntu terminal ssh chris at dc.domain No keys, just typing the password when prompted. The only odd thing

I can't seem to get the "r" modifier to work on inbound SIP calls. The way I understood this to work is that the channel would be answered, and a ring "tone" would be played to the channel. This is not very friendly in that it doesn't honor connection supervision rules, but... who cares? There are some instances where it may be in my interests to get a