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Hai, Very important things are missing here in this one.. smb.conf Samba from sources or os provided? In case of last, centos 7.1 to 7.4 is samba 4.1.12 to 4.6.2 https://wiki.centos.org/Manuals/ReleaseNotes/CentOS7 Various packages have been rebased. Some of those are openLDAP, samba, .... And yes, as Carlos suggested, do check your firewalling also. Maybe samba didnt change but

Dear ALL, I fear my question has already been answered many times before, but I haven't fund that in archives... I am working on spatial datasets and, in most arrays I'm handling, there are two columns dedicated to (x,y)-coords. For different reasons - notably to draw image() plots, I need to have these two columns sorted in increasing order. But sort() and order() seem to apply to

----- Original Message ----- From: yyq at eisoo.com Sent: Tue, 8/16/2011 12:06pm To: samba at lists.samba.org Subject: [Samba] Directly replace smb. [Scanned] Hi,all, I have a redhat Linux of 2.6.24 kernel(x86_64), which smb is 3.0, I want to upgrade it, but the system doesn't have gcc development and no rpm, that means I can't upgrade the smb on this machine.So I compile

Hi all, I have Asterisk 1.2.10, zaptel 1.2.7, spandsp-0.0.3pre22 compiled, and a Sangoma A101, and when I make a call I get this: Jan 8 13:04:06 DEBUG[12252]: chan_unicall.c:2000 unicall_exception: Exception on 19, channel 1 Jan 8 13:04:06 WARNING[12252]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 <- 1101 [1/ 40/Seize /Idle ] Jan 8 13:04:06 WARNING[12252]:

Здравствуйте lartc-request, Friday, October 03, 2003, 8:44:37 AM, you wrote: lrmdn> Send LARTC mailing list submissions to lrmdn> lartc@mailman.ds9a.nl lrmdn> To subscribe or unsubscribe via the World Wide Web, visit lrmdn> http://mailman.ds9a.nl/mailman/listinfo/lartc lrmdn> or, via email, send a message with subject or body ''help'' to lrmdn>

Ok, I thought the point of asterisk@home was that it automatically detected the X100P board and configured it correctly. Is this incorrect? You still need to modify /etc/zaptel files? And not just using the AMP configurator. There is no mention of this on the Asterisk@home webpage. Can anyone who has actually used ast6erisk@home confirm this one way or the other? Thanks, Dean

Peter Thank you for the information. I accidentally deleted Ken's post without having read it. Ken' s thought is great but as you said awful to implement I thought that capture.output would come in handy some time when I first saw it on an unrelated reply. Just thought :- the latex listings package may have alternatives If I remember correctly it has wrapping and other goodies but I

I see. My CPU is a general Core i7 Ivy bridge CPU. On Fri, Apr 10, 2015 at 6:48 PM, Bruce Hoult <bruce at hoult.org> wrote: > I suppose that depends on your CPU. Do you even have a CPU that supports > operations on <N x i1> as packed bits in vector registers? > > > On Sat, Apr 11, 2015 at 12:43 PM, zhi chen <zchenhn at gmail.com> wrote: > >> Yes, but my

I'm working with GXP2130. About 12 phone on gigabit with PC after phone. With Vlans on CISCO switch is stable and not so difficult. This configuration running without problems since July 2013. Quoting jg <webaccounts173 at jgoettgens.de>: >> I have a customer looking to deploy about 20 Grandstream GXP2140 >> phones. Normally they would deploy Yealink brand phones but

Hi, I have a bunch of SIP phones(behind NAT) registering on my * box. I want to find out when they register and de-register. I also want to operate on it, so when they register/de-register, I want to insert calldate into a mysql DB, etc..... Maybe this will help me when, for instance a user tries to register but has the wrong username/password. Now I am aware of regcontext, but it only

On 3/21/2018 7:32 AM, David Minard via samba wrote: > Thanks Carlos, > > The thing is, that I did not upgrade the version of Samba - that is the next step, so the ports used would not have changed. I only updated the OS. > > >> On 21/03/2018, at 10:04 PM, Carlos Alberto Panozzo Cunha <carlos.hollow at gmail.com> wrote: >> >> Hi, >> I have same problem

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

okay, im far from a pam expert, but i don't see any mention of winbind there? It's my weekend at the moment so i can't get to my test box at work to show you my pam module using winbind, but maybe you should check out this page on my website, it's using ldap try and use this and replace any mention of ldap with winbind http://www.yourhowto.org/content/view/35/9/ or i have a

Thanks Carlos, The thing is, that I did not upgrade the version of Samba - that is the next step, so the ports used would not have changed. I only updated the OS. > On 21/03/2018, at 10:04 PM, Carlos Alberto Panozzo Cunha <carlos.hollow at gmail.com> wrote: > > Hi, > I have same problem after update for samba. > I allow new ports in firewall. > >

People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?

I cannot operate lattice plots, notably those in package gstat, within for() loops. (I can neither read such graphics included in a .R (or .txt) file with the source() function). However same commands executed one by one don't fail. Could you please help me find a means to execute lattice plots inside a loop ? Jacques VESLOT CIRAD -----------------------------------------------------

Hi, I had previously posted about connecting an AS5300 to * via SIP/H323. I got it to work via SIP, but only 1 call at a time would work, and if a user from the * side hung up, the cisco would'nt catch the hangup. I an now trying to hook up to the cisco via E1, with a Sangoma A101 card in my * box. I would like it such that I call from * via E1/PRI to the cisco, and call out via R2 to

Hi all, I realize this is OT. I just got a Cisco AS5300, and I need to configure it like such: Asterisk -----(H323/SIP)------> Cisco ----- (E1/PRI)------->Telco So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go out H323 or SIP to Cisco, where they go out PRI. I have the Asterisk side sorted :) (either H323 or SIP), I need help in the

Regards, -- --[ UxBoD ]-- // PGP Key: "curl -s http://www.splatnix.net/uxbod.asc | gpg --import" // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: uxbod at sip.splatnix.net ----- "Michael Graves" <mgraves at mstvp.com> wrote: > Just about anything bootable will

Hello All, I have 4 X100P cards. I was hoping to have card (line) go to separate ext. i.e. Card 1 (XXX)555-0001 My Ext Card 2 (XXX)555-0002 Wife's Ext Card 3 (XXX)555-0003 Fax Ext Card 4 (XXX)555-0004 My and Wife Ext. This is what I have now and all incoming line rings this one extension. exten => s,1,Dial(SIP/300,10) So what is "s" . Thanks, David -- This message has been