similar to: windows 7 professional Samba share access denied

trying to set up a totaly anonymous share for users on a linux box with samba 3.0.10 point to a dir on my hard drive that looks like drwxr-xr-x 2 root root 4096 Jul 7 22:39 sambashare in my smb.conf, for this dir, I have [globalshare] path = /usr/local/share/sambashare read only = No guest ok = Yes guest only = Yes guest account = nobody writeable = yes windows users can

here is what I have: exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN:-1}) exten => _9XXXXXXX,n,Dial(SIP/SIP-Provider/${dialnumber},80) not having success; "Got SIP reponse 503" Service Unavailable" On 04/27/2015 02:19 PM, Bryant Zimmerman wrote: > Motty > Yes > From your dial plan accept 9 + 7 digits

On 10/28/2015 06:37 PM, Pete Mundy wrote: > Hi Motty, > > Isn't the whole point of the nonce in a SIP registration to ensure the > secret doesn't go on the wire in plain-text? Is this not enough, or > are you looking to hide the username too? > > (if so, fair 'nuf, just wondering why :) > > Pete > > Ps, if so then I think TLS is the missing part of

> On 27Apr, 2015, at 16:39, Motty Cruz <motty.cruz at gmail.com> wrote: > > forgot to mentioned I am running Asterisk 1.8.22.0 on CentOS. > > Thanks, > > > On 04/27/2015 02:38 PM, Motty Cruz wrote: >> here is what I have: >> exten => _9XXXXXXX,1,Set(l_HomeAreaCode=381) >> >> exten =>

Hello, I used this guide, it worked for me: http://www.binaryheartbeat.net/2014/03/asterisk-pin-based-dialing.html Thanks, On 07/06/2015 04:54 PM, John Kiniston wrote: > The Authenticate application will do this for you. > > https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate > > You can either give it a single PIN to use for all calls, Authenticate

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.cruz at gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:

On Wed, Jan 25, 2017 at 16:00 Olivier <oza.4h07 at gmail.com> wrote: > What did you exactly upgade ? Asterisk only ? Asterisk and OS ? > How did you installed Asterisk 1.8 and 13 ? From source or from package ? > > I would be curious to see what would happen after downgrading back to 1.8. > > 2017-01-24 21:03 GMT+01:00 Motty Cruz <motty.cruz at gmail.com>: > >

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

Hello, I would like to encrypt password between Asterisk servers and clients. is there an easy way to do so? I am running Asterisk 1.8.22.0 built on CentOS 6.3 Thanks, .Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150729/68b59f7c/attachment.html>

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

Hello, I hope to be in the right list. my question is the following. I can''t seem to get multipathing configure on XCP 1.6. In XenCenter I was able to enable "Multipath" however, in the LUN under general tab I see " multipath 1 of 1 active", the the CLI I do the following command xe --m node session it list only one session. Can someone help me please? Thanks in

On Tue, 28 Apr 2015 07:21:12 -0700 Motty Cruz <motty.cruz at gmail.com> wrote: > here is what I did and worked for me: > > exten => 1381+NXXXXXX,1,Set(CALLERID(number)=3817383444) > > exten => 1+NXXNXXXXXX,2,Dial(SIP/SIP-Provider/${EXTEN:1},80) I find it hard to believe this is working. First, you don't have a leading underscore on your patterns. Your users

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was

Hi, I need a small information on the way rsync copies the files . I am writing the files , links to a list file and I am passing the list file to rsync for copying to remote machine. The contents of the list file (passed to rsync for synchronisation) are /folder1/folder2/rep/01234018329/FSG2_PULL.txt#79GA91205KAE --- actual file /folder1/folder2/users/01234002360/my/FSG2_PULL#79GA91205KAE ----

Hello, I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLS authentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authentication enabled? Any suggestions? Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is there a way to strip +1 out of caller ID? -- Thanks for your support, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141002/6110a4b0/attachment.html>

Hello, I am looking for CDR reporting solution? Any suggestions? I am using Asterisk 13.13.1 I would like a report on number of calls per extension. Thanks, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170322/cb7cd66f/attachment.html>

Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers I was getting this : [Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf after fix global issue

Thanks for your reply, [globals] AREACODE=381 [outbound] exten => _NXXXXXX,1,Dial(SIP/SIP-Provider/1${AREACODE}${EXTEN},80) did not work for me, any ideas? Thanks, On 04/27/2015 01:59 PM, Phil Reynolds wrote: > > > On 27 April 2015 21:32:42 BST, Motty Cruz <motty.cruz at gmail.com> wrote: > >Hello, > > > >I would like to add area code if clients dial 7