search for: anounces

...s being called from a specified location. For that, the command: exten => _107.,1,Dial(SIP/SIPGW/${EXTEN:3},30,A(Anounce.gsm)) should let the called person hear Anounce.gsm as soon as he/she answers. (Only calls with prefix 107 are given this notice). The call proceeds fine, but no one hears AnounceSPF.gsm. I tried putting this file in every sound directory, but no luck. Has anyone used this feature ? Are there any additional parameters or restrictions ? Regards, Jorge A.

I am attempting to setup a asterisk server to take place of my current service with freedomvoice. With the current system a auto-attendant picks up and they go through all the normal menu stuff, once they select the department they wish to speak to the attendant asks them to say their name. Once they do that the system attempts to contact a agent and when that agent picks up the

Hello, we are using asterisk in combination with the voicemail system. I?m just wondering if it is possible to switch the voicemail to an "I am on holiday" mode. This means that the unavailable message is played to the caller but no possability to record a message. So far I did not find an option in the voicemail.conf for this. Any ideas except creating my own ivr menu ? best

In a tearful announcement, Ronaldo,one of the world's greatest soccer stars, revealed his decision to retire, ending one of the most successful careers in soccer. Ronaldo Lu?s Naz?rio de Lima,commonly known as Ronaldo, is a Brazilian former footballer who last played for Corinthians. Ronaldo was one of the most prolific scorers in the world in the late 1990s and the early 2000s. He won his

Hi, does anyone know if it is possible to implement a samba-Setup using Domain users without the use of Winbind? If've the following environment: - Ldap-Server with every Userinformation (Single Point of Administration) - Group and User (also the Mappings and the Passwords) are replicated into the ADS Domain - The ADS is needed to get group Policies working. - The fileserver is Linux with

Hi all, i''ve coded htb-gen, a GPL htb frontend and much more... htb-gen is meant to be an easy, scalable, yet powerfull, bandwidth management tool. You can set up/down portions of bandwith for each host or network, that goes trough your router/firewall. Prioritary traffic(web, mail, gaming, ftp, voip, streaming) is preferred over Junk traffic(kazaa, emule, etc). Also dynamic bandwith

I saw in the anouncement for the upcoming Boston, MA conference that there will be demos of CentOS on an ARM7 system. I am wondering: does this mean that there is somewhere out there a build of some version of CentOS that might run on a Raspberry Pi (which has an ARM7 processor)? -- Robert Heller -- 978-544-6933 Deepwoods Software -- Custom Software Services

Am 04.09.2018 um 09:41 schrieb Sami Ketola: > imapsync always loses data never saw this, be carefull by anounce such myths Best Regards MfG Robert Schetterer -- [*] sys4 AG http://sys4.de, +49 (89) 30 90 46 64 Schlei?heimer Stra?e 26/MG, 80333 M?nchen Sitz der Gesellschaft: M?nchen, Amtsgericht M?nchen: HRB 199263 Vorstand: Patrick Ben Koetter, Marc Schiffbauer Aufsichtsratsvorsitzender:

Hi all, is acl ( with acl plugin enabled ) anounced in imap_capability list by dovecot i cant find it in telnet tests, so is it my fault or is it default beavior using 1.1.rc12 -- Best Regards MfG Robert Schetterer Germany/Munich/Bavaria

Am 04.09.2018 um 16:52 schrieb Sami Ketola: > > >> On 4 Sep 2018, at 17.47, Robert Schetterer <rs at sys4.de> wrote: >> >> Am 04.09.2018 um 09:41 schrieb Sami Ketola: >>> imapsync always loses data >> >> never saw this, be carefull by anounce such myths > > > It is a fact. Imapsync works over IMAP only and IMAP protocol does not even

A friend and I have recently added the ability to announce the callers position in the call queue every x seconds.. or even just inject an anouncement every x seconds. All setup in queues.conf and can be setup per queue. My next project is to add the ability to announce the callers estimated wait time. I want some feedback to see whats the best method to calculate that? What do you want just

Hi, I'm trying to get capiECT working. I'd like to transfer call to another ISDN router connected extension and free channel from router to Asterisk. I get incoming call on CAPI and would liek to transfer it to dialed local extension - 400 in this case: [outbound-capi-local] exten => _4XX,1,NoOp(Transferring to local PBX ISDN number ${EXTEN} on msn CAPI/${CALLERIDNUM}) exten =>

Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so

Hello list, I have a question here that may be a little bit strange for some of you. I would like to send an INVITE from Asterisk to a given client without any SDP anouncement in it. Indeed, that is pretty useful for Click to call applications for instance, where you have no way to know which codecs are supported by the client you try to reach. Moreover, you let the client decide the

Hello everyone, Last week the CVS version of tinc gained the ability to run under Windows, in a Cygwin environment (http://www.cygwin.com), making use of the virtual network driver from the CIPE project (http://cipe-win32.sourceforge.net/). Preparations were also being made to support Windows without Cygwin. Today James Yonan of OpenVPN (http://openvpn.sourceforge.net/) anounced a port of

Am 04.09.2018 um 17:18 schrieb Sami Ketola: > > >> On 4 Sep 2018, at 18.00, Robert Schetterer <rs at sys4.de> wrote: >> >> Am 04.09.2018 um 16:52 schrieb Sami Ketola: >>> >>> >>>> On 4 Sep 2018, at 17.47, Robert Schetterer <rs at sys4.de> wrote: >>>> >>>> Am 04.09.2018 um 09:41 schrieb Sami Ketola:

>I went back to the example system direct from CVS with small >additions to sip.conf and extnsion.conf needed to make one >xten X-Lite phone work. I can dail in and hear the anouncements, >call the demo server at Digium. The audio quality I hear >comming from Asterisk back to X-Lite is good (9 on a 10 scale) >but the sound volume comming from the X-Lite extension is very low

Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early

Am 20.03.2015 um 10:45 schrieb Volker Lendecke: > On Thu, Mar 19, 2015 at 05:47:11PM +0100, Robert Schetterer wrote: >>> Well the API is propably just stuffing blobs into extended >>> attributes directly from userspace. That's how most of >>> the NFSv4 ACLs usually get done :-(. >>> >>> Of course all implementations use different blobs containing

Hi , i got trouble with master login thunderbird anounces no INBOX there at master login also telnet looks strange normal user login 1 login ...... 1 OK [CAPABILITY IMAP4rev1 LITERAL+ SASL-IR LOGIN-REFERRALS ID ENABLE SORT SORT=DISPLAY THREAD=REFERENCES THREAD=REFS MULTIAPPEND UNSELECT IDLE CHILDREN NAMESPACE UIDPLUS LIST-EXTENDED I18NLEVEL=1 CONDSTO...