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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...

...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 pr...

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 spik...

...nbind, 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 30 21:44:56 myserver pam_winbind[7349]: user 'stile' granted access Nov 30 21:45:44 myserver sshd(pam_unix)[7349]: check pass; user unknown Nov 30 21:45:44 myserver pam_winbind[7349]: user 'stile' granted access Using Red Hat EL AS 3 + samba-3.0.9-1 + krb5-lib-1.3.1 /etc/pam.d/sshd #%PAM-1.0 auth required pam_stack.so s...

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 l...

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 th...

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 affordab...

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'...

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_NUMB...

...ould i try another method? What is the next best setup method? I am left wondering what the best options are available at this point, as things seem hopeless. I have followed steps outline in Samba-3 By Example, by John H. Terpstra, chapter 9.3.3 -- ._____________________. | \0/ John Stile | | UniX Administration | | / \ 510-305-3800 | | john@stilen.com | .---------------------. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url...

...r/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. ------------------------------------------- scp -v text.txt jstile at HPUXserver:~/test.txt Executing: program /opt/openssh2/bin/ssh host HPUXserver, user jstile, command scp -v -t ~/test.txt jstile at HPUXserver's password: Permission denied, please try again. jstile at HPUXserver's password: ksh: scp: not found lost connection

...gt; Password for vivek at SQUID.BIZ: > # > > Anything need to be modify on the Windows side ??..next step i will remove > the system from the domain and try everything... > > Thanks in advance. > > Regards, > VIvek > > > > On Mon, Nov 15, 2010 at 8:25 AM, John Stile <john at stilen.com> wrote: > >> "Invalid credentials" points to a problem, thought I'm guessing, with >> the domain membership. >> >> I'm really not sure what it means. >> >> Does 'ads testjoin' show anything? >> >>...

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 S...

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