search for: stiles

Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to w...

...ing? sip / PRIs ? what load? > 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) > On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: > > Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: > > hi, > > i have strange pro...

Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> hi, >> >> i have strange problem with asterisk 13 mixmonitor, recording to wav >> (centos6) >> when the system is under load, there are sometimes missing syllable >> >> there arent BIG spike...

Samba setup as a Member Server in native AD domain with winbind authenticating AD users for access to shares. My understanding is that with pam and winbind, domain users can log into the samba server via ssh, even if they do not have a local user account? Logs shows access granted but user unknown, so I must be missing something and need some help. /var/log/messages during an ssh login: Nov

Hi AJ, Am 06.06.2016 um 10:14 schrieb A J Stiles: > But why not call an AGI script, have this check the caller ID against a MySQL > database and return a status -- blocked or not -- in a variable? Then you can > manage individual number blocking in a much cleaner, more extensible fashion. > > Feel free to ignore me if it sounds li...

Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26, passing it to SIP/708708-000010b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26,

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > On Monday 06 Jul 2015, Luca Bertoncello wrote: >> Well, but for voice quality, which codec is better? >> alaw or gsm? > > A-law is better for voice quality (sorry, thought my original > explanation was > obvious). But note...

Zitat von A J Stiles <asterisk_list at earthshod.co.uk>: > Yes. You should definitely be using A-law for calls to the Outside World. Well, I wanted to change these settings, but I'm not sure, where I have to do that... I think in the users.conf, but I think, the "allow" keywords is for the...

I use winbind+AD for single sign-on to many Linux machines, and the Linux hosts automount home directories on one Linux file server. I am faced with a need to synchronize the *.tdb files on the file server with all the Linux machines for consistent UID-to-loginID mapping. Has this been tried by sharing the /var/lib/samba/ over nfs with the other servers? Are there any pitfalls? Has this been

>>>>> "AS" == A J Stiles <asterisk_list at earthshod.co.uk> writes: AS> If you are paying for a business-grade Internet connection, you AS> should get a static IP address -- or a block of them -- as AS> standard. Maybe you need to change your ISP? In some places (including here) static ip is not affordabl...

On Wed, 10 May 2017, J Montoya or A J Stiles wrote: > Presumably your staff carry mobile phones. What about an app that gets > the ID of the cell tower to which it is connected, and passes it and the > SIM number in a HTTP request to a server you control? The problem is that they are supposed to use the 'site landline' t...

On Wednesday 10 Jun 2015, Luca Bertoncello wrote: > I'm very sorry to write that, but these answers are really NOT helpful... > I searched two days long how can I check it and didn't found anything > useful... > > Could someone suggest me a way to check if my Asterisk is an "Open > Relay" that accept connections from every peer? Someone on this list is bound

Is anybody else using the following combination: * a TE410P card (wct4xxp driver) * a BT ISDN connection * DAHDI 2.3.0.1 * Asterisk 1.6.2.9 I'm trying to configure a new box to replace a legacy system (same hardware; some old version of Asterisk with Zaptel; works lovely but hopelessly out-of-date) and not having much joy. Specifically, I couldn't get it to see a D-channel on

On Monday 20 Dec 2010, Olivier wrote: > 2010/12/20 A J Stiles <asterisk_list at earthshod.co.uk> > > > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application > > (written by someone else before me) which sets up calls by creating > > files of > > the general form > > > > Channel: SIP/$INSIDE_NUMBE...

Samba is trying to be a member server in an AD in native mode, using winbind, nss, and kerberose. There are 3 kdc's (2 are Win2003, 1 is Win2000), samba server is RH-AS3 + Samba version 3.0.9 (from samba.org) + krb5 1.3.1-6 (from Fedora Core). I thought I had things working (join succeeded, could access shares, modify files), and then it stopped working. After clearing out the host account

I upgraded a compiled version of ssh-1.2.27 to a swinstalled depot of OpenSSH_2.5.1p1 on HPUX-11.00. I created links in /usr/local/bin/<ssh program> pointing to /opt/openssh2/bin/<ssh program>. Ssh works. Scp does not. HP support does not support ssh. Below the line you will find the output of a verbose scp command from the server to it self.

Hi John, The same smb and winbind configuration ( same SUSE box ) works good other Windows AD servers. "#wbinfo -u" and "#wbinfo -g" returns the users and groups respectively. Thanks for your great help !!! what is the difference between "#net rpc" and "#net ads" ?..if you have time, give some explanation.. Regards, Vivek On Mon, Nov 15, 2010 at 6:56

Dne 4.2.2016 v 12:17 A J Stiles napsal(a): > On Thursday 04 Feb 2016, Marek ?ervenka wrote: >> hi, >> >> is there way to get SQL schema for Asterisk 13.7.0 without alembic? >> thanks > Assuming you already have Asterisk up and running, you can just use > > $ mysqldump -d -uroot DATABASE TABLE1...

...ou either need the phone to be answered, or the phone to be answered and and the challenge entered. Adam Goldberg AGP, LLC +1-202-507-9900 -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of J Montoya or A J Stiles Sent: Thursday, May 11, 2017 7:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?) On Wednesday 10 May 2017, Steve Edwards wrote: > On Wed, 10 May 2017, J Montoya or A J St...

Hi, I need to make calls to a list of numbers one at a time and once the user pick the phone connects to an IVR where I can get few data, after a call finishes the 2nd number get called and so forth. I'm familiar with Asterisk/Elastix but the Campaign feature on Elastix does not seem to fill this need. I'm now looking GoAutodial & AsterCC. Anyone with an idea to solve this issue I