similar to: Asterisk 1.6.2.13 Audio Prompts Stopping

On Tue, Jun 30, 2020 at 11:24 AM Rowland penny via samba <samba at lists.samba.org> wrote: > > On 30/06/2020 09:50, Anders ?stling wrote: > > >> You have 'workgroup = HPLTS' and 'idmap config dg11', again, they must match > > As I wrote in the previous reply, that was a mistake from the initial > > deployment. However, I have a copy of the VM and

If anyone has time to look at this problem I would appreciate it. I think I am looking for a way to increase the "timeout" in samba (in the smb.conf file) for reporting a "down" link or "can't read xyz file". However I am writing to you folks because this problem has come up while using rsync my problem ========== We have a wan in which links from a

Hi, I have an strange issue on our company network. We run samba4 ad-dc's on four branches as separate sites, they are connected via ipsec tunnels, all servers are debian wheezy systems using sernet 4.1.9-8 samba packages. We use roaming profiles with folder redirection configured via GPo's. In tree of the four branches users suddenly losse the connection to their home shares, since

On Tue, Jun 30, 2020 at 11:57 AM Rowland penny via samba <samba at lists.samba.org> wrote: > > On 30/06/2020 10:34, Anders ?stling wrote: > > On Tue, Jun 30, 2020 at 11:24 AM Rowland penny via samba > > <samba at lists.samba.org> wrote: > >> On 30/06/2020 09:50, Anders ?stling wrote: > >> > >>>> You have 'workgroup = HPLTS' and

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

---------------------------------------- From: "Andrew Latham" <lathama at gmail.com> Sent: Thursday, January 17, 2013 3:04 PM To: bryantz at zktech.com, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] Conf Bridge On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman <BryantZ at

The Cisco/Linksys SPA devices are also able to be provisioned automatically. On Wed, Apr 15, 2015 at 3:20 PM, Bryant Zimmerman <BryantZ at zktech.com> wrote: > Alejandro > > All of the Grandstream devices can be remote provisioned if you know what > you are doing. > > Bryant > > ------------------------------ > *From*: "Alejandro" <cdgraff at

Hi, I can cross compile asterisk 1.4.21 on arm (imx27) using ltib I want to cross compile the new version 1.6.2.13 but there is an error when I execute the commands : ./configure --build=i686-pc-linux-gnu --host=arm make menuselect The configure seems ok, I have the result info : *configure: Package configured for: configure: OS type : none configure: Host CPU : arm configure:

I have the following senerio. Call file calls 1st party. When connected give called party option to connect to second party. Issue Dial to second party. Caller answers and the two are bridged together. My issue is that 4 out of 5 calls fail to bridge the audio. Am I missing something or is there some kind of bug? Here is my test dialplan ;Dialer Base Code Files. ;Variables

The Asterisk Development Team has announced the release of Asterisk 1.6.2.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files.

The Asterisk Development Team has announced the release of Asterisk 1.6.2.13. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ This release resolves an issue where the .version and ChangeLog files were not updated for 1.6.2.12. Asterisk 1.6.2.13 has no additional changes from 1.6.2.12 other than the .version, ChangeLog and summary files.

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

I did an upgrade to the SVN trunk on the 12/9 and when I looked in my mysql table for CDR's today there are no entries since the update. I have rebuilt and re-installed and re-started asterisk still no CDR's flowing to mysql. I did not change any configs. I checked to make sure that the cdr_mysql option was selected under the make menu options. The module shows it is there when I do a

Is there a way to limit the items returned by pjsip show [type] using like chan_sip allowed for sip show peers like xxxx, but I can't seem to figure out how to lookup or limit my returns with pjsip Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

I have a project that is requiring the use of MS SQL from asterisk. I get an error when the pjsip contact tries to update the contact table. [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018: [FreeTDS][SQL Server]Conversion failed when converting the varchar value '3.000000' to data type int. (101) The datatype

We have written some monitoring and stat collection scripts that use asterisk -rx "command" The script runs once a min and logs data and posts any critical notifications. Everything is working well with this method but we get the -- Remote UNIX connection / disconnect message once a min and we would like to suppress it. Is it possible without reducing the verbose logging level.

Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input.