search for: mundy

...nd 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 unfortunately none of our SIP > phone have the facilities to use multiple line and UI. > > I can see incoming numbers on my softphone(Zoiper) when a incoming call > hits. I liked your incoming caller ID customize idea. > > Is it...

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/1...

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

...s 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 nerdvittles: http://mundy.org/blog/index.php But I'm not sure that this actually does concurrent calls. I'm thinking my best bet is writing some fast agi to parse a mysql database, then create call files. Use asterisk manager interface to monitor calls and that way I can keep the preset concurrent limit. Any ideas...

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 :) > > P...

...015, at 4:11 PM, Thyda ENG <engthyda at gmail.com> wrote: > I mean by sending the .jpg, or .png or ..... file. > > On Tue, Aug 25, 2015 at 11:10 AM, Thyda ENG <engthyda at gmail.com> wrote: > 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...

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:

...-- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Joseph Tanner Sendt: 4. februar 2006 11:51 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users] callback script? This is what I use, more or less: http://mundy.org/blog/index.php?p=73 , go down to "Incoming Call Context" (about 1/3 down). I had to modify it a bit, as I actually need Asterisk to pick up and listen to some DTMF digits before hanging up and calling me back, but it works great for me, and requires no external agi scripts. Joseph T...

Hey all. I am working on getting solaris 10 and 9 clients authenticating to active directory. We have a test setup with windows 2003 r2 and the unix fields filled out. All was going well with a combination of pam_krb5 and nss_ldap and ldap mappings, including working automouting of home directories. This included orginally, set up instructions from microsoft where they said to create a user

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

Hi! I've set up a Samba PDC(3.0.4+LDAP) and attached a W2k client to it. I've created users and group. I can log on to the W2k client. Now to my problem: I wanted to add a user to the local filesystem(for testing). So I opened up the Security Tab and listed my domain. Lo and behold. No users. That strange I thought and added the user manually and hit the Check button. It worked

-- jim knoble | jmknoble at pobox.com > On Apr 30, 2017, at 19:40, Darren Tucker <dtucker at zip.com.au> wrote: > >> On Mon, May 1, 2017 at 12:09 PM, Damien Miller <djm at mindrot.org> wrote: >> >> Hi, >> >> I just deleted SSHv1 support in OpenBSD and portable OpenSSH. There's >> probably a little dead code still to be expunged, but

> 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 s...

...[2004/02/09 14:51:27, 5] auth/auth.c:check_ntlm_password(271) check_ntlm_password: sam authentication for user [dario] FAILED with error NT_STATUS_WRONG_PASSWORD [2004/02/09 14:51:27, 3] auth/auth_winbind.c:check_winbind_security(80) check_winbind_security: Not using winbind, requested domain [MUNDY] was for this SAM. [2004/02/09 14:51:27, 10] auth/auth.c:check_ntlm_password(259) check_ntlm_password: winbind had nothing to say [2004/02/09 14:51:27, 2] auth/auth.c:check_ntlm_password(312) check_ntlm_password: Authentication for user [dario] -> [dario] FAILED with error NT_STATUS_WRONG_P...

...er says the same occurs at their end, his/her voice 100% reflecting back to him/her. I usually have to reset the device completely at that point. Other times, I'll get a call and talk for 1/2 hour with no problems whatsoever. I have it set up with a@h V 2.2 according to the setup at http://mundy.org/blog/index.php?p=65 altough I am not using the caller ID portion of his setup. Also I get the following strange error: chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 102 Cmd SIP/2.0 I can't seem to pin it down. I've checked the source (c...

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. > &gt...