search for: mundi

I have added CID name prefix on inbound route. and it works fine :) now I can simply forward five incoming routes to one extension. and as far as I guess, if I add CID name prefix for every number. it should work :) thanks alot :) On Tue, Mar 22, 2016 at 2:28 AM, somsad khan <ctrlz.network at gmail.com> wrote: > hello Pete Mundy, > > thanks alot for your idea and reply. but

Hello all: I upgraded to 1.4.3 last night and use MySQL for CDR. I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that 1.4.2 did not. I use a few macros in my dialplan to handle outgoing calls (lcr type stuff) and in addition to the proper CDR for the call itself I also have records to 's' in the same dest-context and entries to 's' in the default context.

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi, I just deleted SSHv1 support in OpenBSD and portable OpenSSH. There's probably a little dead code still to be expunged, but all user-visible functionality and the bulk of the supporting infrastructure is gone. Sic transit gloria mundi. -d

Many desk phones support multiple simultaneous SIP registrations. You could use BLF buttons for each SIP registration and the operator uses the LEDs as their queue as to which account is ringing. Alternatively the phone's UI may be able to indicate which account is ringing without the need for BLFs. Another option is to re-write the CALLERID(num) or CALLERID(name) to indicate the inbound line

Hi, I'm helping out with a political campaign and would like to use asterisk to blast out about 200,000 calls with a short message from the candidate. Provider: I'm thinking voipjet may be a good solution? Hardware setup: I will have access to several T-1 lines so I would just want to set up the dialers to limit the number of concurrent calls and so forth. I found teleyapper on

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

Hmm, most phones I've used wouldn't have the capability of displaying a bitmap image due to only having minimal monochrome displays. What sort of end device do you perceive to display these images? Can you give links to any devices with support for such things? I'm assuming you mean only a still-photo image, not video image. Perhaps you could use a video channel for this and simply

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:

Thanks. I'm able to getting the asterisk calling back to my cellphone. But when I get to the authentication I get this message when I start to dial in my password: NOTICE[5178]: rtp.c:509 ast_rtp_read: Unknown RTP codec 96 received Is this a DTMF failure of some sort? Thanks again. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com

...but to no avail. They keytab also appears to have entries for both, so I don't understand this error. DNS works in both directions for this host, and like I said when creating a keytab from a user account on windows, manually, this process works. If I run something like "kinit -k host/mundi@TESTAUTH.NETWORK" I get a preauthentication failed message. If insted of the mundi I make that fully quallified I get the client not found in kerberos database error. I should also mention in order to get this far I had to add supported_enctypes = des-cbc-md5 and various other lines to krb...

Hi, Sorry to insist, but I would really like to be able to use Xen and my zaptel hardware all together in Dom0. I was just wondering if the 3.1 release could contain some changes compared to 3.0.4 related to IRQ handling or scheduling that could workout my problem. Thanks, François. ---- I actually first asked to asterisk mailing lists, and a few persons told me that it was Xen''s

...oups was visible. What am I doing wrong? Can anyone give a hint or two on where I should go? Mvh, Roger Martensson E-Mail: Roger.Martensson@mh.se Work: +46-(0)63-16 54 45 Cell: +46-(0)70-629 54 45 MSN: Roger.Martensson@mh.se NetMeeting: Intranet only And dont forget: Magister Mundi Sum

...t;> Hi, >> >> I just deleted SSHv1 support in OpenBSD and portable OpenSSH. There's >> probably a little dead code still to be expunged, but all user-visible >> functionality and the bulk of the supporting infrastructure is gone. >> >> Sic transit gloria mundi. >> > > Quamvis ejus problems, est etiam a magnus lenimentus super eam telnet. Ut cum multa -- linguam latinam, exempli gratia -- pugnatum est multo longior quam exspectavimus. Ni esperas, ke la dua provo renkontas pli bonan finon. -- jim knoble | jmknoble at pobox.com

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no >

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

On 2017-04-19 02:39 AM, Pete Mundy wrote: > Hmm... Above my pay grade I'm afraid! Looking at your 'voicemail > show users' I can't see why the vm_authenticate function is > failing to read the username :( I can answer that one. It's because we can't enter 'stocktrans2' from a telephone so we just hang up. The question is, why does it ask for the

I have really weird problem with XP Pro and Samba 3 (I tried 3.0.1 and 3.0.2). Samba 2.2.8a works with no problem. I can login with no problems. But when I login and logout few times (sometimes one, sometimes five times) then XP can not load roaming profile from the server (login still works). I can access to all shares with no problems. The only way to fix that is to restart XP machine. And then

To All, I recently purchased a Sipura 3000 for home use. Generally it seems to work fine, but it also majorly screws up on a frequent, almost daily, basis. Terrible echo problems that come and go, and sometimes it just seems to "lose its way" and can't re-register. For example, I'll be talking to an incoming caller and an echo starts quietly in the background. Within a minute

Yes, I mean sending image file. On Tue, Aug 25, 2015 at 10:56 AM, Pete Mundy <pete at fiberphone.co.nz> wrote: > Thyda, > > The term 'image' can be quite ambiguous in computing. For example you > could be referring to a firmware image for a phone or you could be > referring to some form of live video channel support. Or something else. > > Can you be more