similar to: Asterisk / FreePBX Support / Reseller

Size: - one location - 15 IP Phones ( 1 dect) - Create new voip trunk (current are ISDN) (30 number block) - LTS is important - an SLA is optional at the moment there is no one On 11.12.2017 22:31, Ron Wheeler wrote: > You might want to add some details > - size of the project > ?-- number of locations > ?-- number of extensions > - are you converting your trunks? > - what are

I know but this is not my sole decision. On 12.12.2017 16:17, Ron Wheeler wrote: > If your phone system goes down and you can not get it back up until > tomorrow afternoon because your support person is on another project, > you may wish you had an SLA.

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: > both phones are in the same net. > when the soft phone is shut down, on hardware phone only an led is > flashing to show an incoming call but no sound. > > both phones use the same extension. that is the reason why I use pjsip. > > On 06.02.19 14:59, Antony

that was my first idea. and how should an other user know which number he should dial? user a: soft phone extension 100 hardware phone extension 101 On 06.02.19 15:25, Mitch Claborn wrote: > You can do this in the dial plan. Register the devices separately and > include both addresses in the Dial() command. > > > Mitch > > On 2/6/19 8:16 AM, basti wrote: >> In

On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: > These two phones are not using the same extension, are they? If you shut down the softphone, does the hardware phone then ring? Antony. > On 2/6/2019 8:49 AM, basti wrote: > > both phones are registered. and the hardware phone can also make calls. > > but an incoming call is not displayed and also not hearing.

Hello, I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set "Max Contacts = 2" to have register both at the same time. But Only the softphone is ring. the hardware phone is mute. How can i fix this?

Hello, I've been tasked to look for ways to resell to others the service that one of a trunk provides.. In other words, i want to configure my current Asterisk (Ver. 1.4.26.1) with Freepbx 2.6.0 so i can act as a trunk to others.. I would provide an IP to them from one of my servers and they will use that IP to connect to me and i will connect them to my trunk/provider. If

Hallo, is there a Freepbx mailinglist? or can this be posted here? Best Regards,

Hi List, does somebody know a SIP Provider which offers reseller possibilities? Moritz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040828/bddb5898/attachment.htm

Hello Ron, Which kernel do you run Asterisk/Freepbx with ? Cheers 2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwheeler at artifact-software.com>: > CentOS 7 works well with Asterisk. > Install latest CentOS7 with updates install asterisk > > I am running FreePBX on CentOS 7. > > Ron > > On 14/12/2017 10:38 AM, Olivier wrote: > > Hello, > > I'm used to

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

I've got a project where I need to sell a voip QOS product from Australia to US resellers. I don't suppose anyone here knows where I can find a list of a whole heap of US resellers do you in either VOIP or IP space? Regards, Dean Collins Cognation Pty Ltd dean@cognation.net +1-212-203-4357 +61-2-8307-3503 (Sydney in-dial) -------------- next part -------------- An

I'm in the Northeast US and looking for any recommendations on Level3 resellers. I don't do enough volume to go to Level3 directly. If there's anybody you'd definitely avoid I'd love to hear about that too. Thanks, Adam

Dear Sir, We would lik to subscribe the support for OpenSSH for a tender project. Please kindly advise the annual subscription plan (with details of SLA) & annual fee. For the price, is it determined by number of server? or number of CPU? or number of core? Shall we get quotation/support from you? or please advise the reseller contact. Since the tender will be closed next week, may we

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

We need to resell systems that run commercial software on Linux. We were using Red Hat ES 3 originally. But we later realize that we cannot redistribute RHES without paying them huge subscription fee because of the trademark. I came across CentOS from WBEL mailing list. This looks like a godsend to me. I have some questions regarding CentOS. I would appreciate it if someone can provide me the

Hello, I'm used to install Asterisk on Debian stable platforms. A customer is asking how I would proceed on a CentOS platform. After a short research (see [1] as an example), I'm wondering what are general kernel practices on CentOS regarding Asterisk and when targeting stability: - Is it recommended to upgrade kernel version(s) (ie moving from linux 3.10 to 4.3) just after OS

I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it