search for: fiberphon

...possible to add company name with incoming numbers. so that company > name or any signal will appear with incoming call numbers, will be easy to > identify by employee that call is coming into which number. > > thank you > > On Tue, Mar 22, 2016 at 2:12 AM, Pete Mundy <pete at fiberphone.co.nz> wrote: > >> >> 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 ma...

...hyda 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 be more explicit as t...

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 expl...

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

If you want to use dahdi dummy driver inside asterisk for timer then this is possible with openvz based container virtualization. We have tested vicidial in this mode for 5-10 agents and it worked well. Mitul Limbani On Apr 8, 2016 8:52 AM, "Pete Mundy" <pete at fiberphone.co.nz> wrote: > List, > > Might as well throw my hat in the ring! > > I can't say it's the 'best' way to do it, but I've been running Asterisk > VMs inside the free 'VirtualBox' software for many years with nill issues > (well, nill related to t...

Heya Rodrigo Not sure, but this expansion on Sammy's concept may help you achieve the delayed ring on the secondary extensions you were looking for. exten => _600.,1,Dial(PJSIP/${EXTEN}) exten => _600.,n,Hangup exten => _600.wait5,1,Wait(5) exten => _600.wait5,n,Dial(PJSIP/${EXTEN:0:4}) exten => _600.wait5,n,Hangup exten => 555,1,Dial(LOCAL/6001&LOCAL/6002.wait5)

On Thu, 31 Aug 2017, Joseph Smith wrote: > So I am looking for a better way to allow several thousand callers to > listen to this IVR menu at the same time. I'm thinking multiple hosts. I'm not a fan of 4,000 eggs in one basket. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com

On Tue, 18 Oct 2016, Pete Mundy wrote: > If you want to know what is _really_ in that file (including all > invisible characters and anything else that wc etc might not count), > pipe it through 'hexdump'. > > cat?/home/test/feature-1.txt | hexdump Or just: hexdump /home/test/feature-1.txt -- Thanks in advance,

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

I mean by sending image by using sip channel just like we can send text message and what about sending image file ? On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp <jcolp at digium.com> wrote: > On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote: > > Dear Sir, > > Kia ora, > > > > > I current have done successfully with sip message over asterisk server , > >

I have tried with xen and kvm both are working fine. On Wed, Apr 6, 2016 at 3:44 PM, Loic Chabert <loic.chabert at voxity.fr> wrote: > Hello, > > Work well with kvm and centos 7. > Some ajustements has to be made with systemd. > > I'm using it in production since 1.5 year now, no issue to report. > > Regards. > Le 6 avr. 2016 21:13, "Yves"

...lp to change the codec that is being used? I am not sure about the MoH but the audio files I am using are gsm. Thanks Joseph ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Pete Mundy <pete at fiberphone.co.nz> Sent: Thursday, August 31, 2017 2:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ERROR during high volume MoH dialplan >> On Thu, 31 Aug 2017, Joseph Smith wrote: >> >> So I am looking for a better way to allow several t...

Am 19.08.2015 um 19:07 schrieb Steve Edwards: > Please don't top post. > > On Wed, 19 Aug 2015, James Cass wrote: > >> Steve, would you be willing to share that "quick bash script"? > > There's no magic in the script, but here it is, embarrassing myself: > > cp sample-call-file /tmp/ > chmod +x /tmp/sample-call-file >

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

Hi, I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see any issues until today. The setup I configured for inbound fax is quite simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38 protocol and later Asterisk stores/forwards the fax to specific end user. The configuration I made in sip.conf for enabling T38 is listed below; t38pt_udptl =

We have a template for extensions and voicmail. They look like this: exten => %ACCOUNT%,1,Verbose(0,Entering extension %ACCOUNT%) same => n(DialDesk),Verbose(0,${CALLERID(all)} Calling ${EXTEN}) same => n,Dial(SIP/%ACCOUNT%,30) same => n(VoiceMail),Set(CDR(userfield)=VoiceMail) same => n,Verbose(0,${CALLERID(all)} going into voice mail for %ACCOUNT%)

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

> 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 >