Hello German descriptopn below: Wee need as soon as possible: Install and Configure Asterisk PBX or Freepbx with VPN and nice Gui: - Configure and Install Asterisk - VPN - Fax2 MAil - Mail2 Fax -Mailboix - Time based Rules - Conference Room - Call Trough - German Lanmguage Files - Call and Waiting Groups - Call RTecording - Announcement
Wir suchen jemanden, der unsere FreePBX-Anlage, die wir angemietet haben, administrieren kann. Sie ist eingerichtet, aber ab und zu sind Ergänzungen oder Änderungen notwendig, wie auch im Moment. Das ist alles nicht zeitkritisch und kann auch außerhalb der normalen Geschäftszeiten gemacht werden. Für eine Beauftragung an ein Unternehmen ist der
...eine Software um CDR (Call Data Records) Dateien von Asterisk Servern zu erzeugen und nach festgeletem Prozess zu übermitteln (siehe Dokumentation). Wir haben Asterisk und Freepbx Server und die Software soll jeden Tag zu 2 Zeitpunkten ein neues CDR zu generieren mit den Daten wie in der Beschreibung (angehängt). Die Übermittlung soll automatisch erfolgen
We need a server specialist that can install tonight (with information from our developer) an Asterisk (with freepbx) VoIP Server on a CentOS. We already hired some previous server guys who did not finish their job. You must do that, when you can do that in a couple of hours we pay you $250. But must finish today!!! Next to installation and proper
...with A2Billing installed on it, so I need someone to write the installation guide commands to establish it completely on CENTOS 6.7 VPS starting with Asterisk, a2billing & freePBX , I will be using it for PC2Phone AND calling card, so the installation manual you write should indicate the proper setup. I know that there are alot of websites that shows
Asterisk has stopped running on my version of Freepbx. I do not know how to access the command line or the commands to restart. I just need someone to show me, preferably by doing a screen share and walking me through the steps on how to 1. access the command line on my server 2. enter the commands to stop, and restart asterisk
I need to install a2billing in freepbx 13 with asterisk 11 and have to verified the command script by using teamviewer in my virtual local server. Kindly also need to show some configuration, like in freepbx IVR has two option and one of them is calling card option. so if customer chose the calling card option then how to divert them to a2billing and
Hi Toshazed, I need your help to install a2billing in freepbx 13 with asterisk 11 and need to show me the command script. it will be install in my virtual local server. Kindly please help me to show other configuration which will be discuss after installation.
...lograr es poder realizar llamadas telefónicas desde el mismo Bitrix a través de las centrales Issabel ya configuradas. El bitrix cuenta con instructivo para conectarlo con FreePBX. Si bien este se maneja también con Asterisk, es necesario hacer la configuración....
I have local FreePBX server with 2 Network Interface Cards eth0: for local network DEVICE="eth0" BOOTPROTO="static" HWADDR="xx:xx:xx:xx:xx:xx" ONBOOT="yes" TYPE="Ethernet" IPADDR="[Zur Anzeige der URL Anmelden]" NETMASK="[Zur Anzeige der URL Anmelden]" NOZEROCONF="yes" BROADCAST="[Zur Anzeige der URL Anmeld...
chan_sip.c:10880 process_sdp: No compatible codecs, not accepting this offer! ... compatible codecs, not accepting this offer! I am redirecting my calls (old provider) to my new freepbx, I have a trunk added but there is an error .
Create a Survey FreePBX/Asterisk module to measure customer satisfaction levels for our call center agents services. Current software versions: Elastix 4 Here is what is needed. After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 1 or more fixed questions. We
I need help to set-up 1 ringgroup to 1 external number. This ringgroup should then use a predefined trunk to make the outgoing call. So when I call the ringgroup I should be connected to the external number using the predefined trunk. This should also work when using inbound routes to a ringgroup. I need your remote help. Please give me a fixed price.
connect to a device grandstream voice ip phone + add sip trunk (outbound minutes) , enable recording all calls and access to them freepbx is mounted , it is working to make inbound calls and outbound calls (with softphone zoiper)
want to set up my very own SIP server /VoIP system with a VPS server i will buy for the project....want to dial 800 type numbers thru MicroSIP, no web interface is needed. I need someone to fully set up the system and explain everything to me. Specifications will be sent when project is accepted.
The project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have
...Windows 10 Enterprise Windows Server 2012 R2 Windows Server 2016 Peplink Router Balance Series Pepwave SOHO series Pepwave Access Points Proxmox cluster Centos OS asterisk FreePBX Arista Switches Nortel/Avaya Switches FreeNAS Cannon MFC8080cw printers Xerox 6605n printers Tripplite UPS with SNMPWEB card installed APC Automatic Transfer Switch AP7750
FreePBX 13 Random trunks MOD Working on a live server with calls, Make modification to the Outbound Routes of FreePBX 13 to select a random trunk for the Trunks sequence list for outbound dialing. Trunk Selection is base on the ratio of MAX channel of each trunk. Please do not bid if you don’t know fully to implement this MOD, If you have any question
So, I have a stock standard, fresh FreePBX installation (in production) I also have a ASP.NET project that has coding in it that pulls call information from the Asterisk database. The ASP app is currently being rebuilt and relaunched and so help is needed with someone in experience in all of the areas mentioned in the project title to put the pieces
Our project consists of two parts: 1) Select the correct customer inside POS automatically wh...printed by Odoo POS, and the system then knows that this delivery man has delivered the scanned orders. We will provide an Odoo v9.0 server where we will also have installed a Freepbx asterisk server along with the OCA connector-telephony plugins installed.
Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDrive App. Need check the requirements from Pipedrive. [Zur Anzeige der URL Anmelden] The Features I need are: Click to Call (CTI) - Click in contact in PipeDrive and call to customer. Call History - All calls made will be logged in the customer details
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be completed in order for push notific...
We'd like to allow a parked caller to press a DTMF which sends the caller to the user that originally parked them. Please provide a full project plan and how much time you'll need to accomplish this.
Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from
I have 2 FreePBX Servers [A & B] Server A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and
Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem