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="[log masuk untuk melihat URL]" NETMASK="[log masuk untuk melihat URL]" NOZEROCONF="yes" BROADCAST="[log masuk untuk melihat URL]...
We need to integrate an asterisk pbx with SuiteCRM if possible with freepbx or raspbx Integration means: - On inbound call open CRM calling contact information - Clicktocall from CRM - Logg inbound and outbound calls in CRM
Do you have interest in conference call service ? Like [log masuk untuk melihat URL] or [log masuk untuk melihat URL] ? Who can make the best portal-experience website including all features of the FreePbx or Kolmisoft platform or mor-platform ? https://wiki.freepbx.org/display/FPG/Conference+Module+User+Guide or [log masuk untuk melihat URL] So in the end we provide
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
I need someone to set up SNMP monitor and Syslog server. Any OS or program is fine. I will do basic install of OS and then you can install software. I need job from start to finish and ongoing support on an as needed basis. There will be HDTVs to monitor system and if the internal web browser in the TVs can be used that would be great. If external computer
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. [log masuk untuk melihat URL] 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.
Freepbx installation and configuration over raspberry Integration with SuiteCRM: popup contact form with incoming call, clicktocall for outbound calls ann log incoming and outgoing calls SuiteCRM is in local server
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
I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?
Basicamente no puedo terminnar de encontrar la configuracion ideal para que el GXW4108 permita enviar y recibir llamadas administrado por una FreePBX , se logro configurar pero inmediatamente se cuelga. Yo creo que son los parametros GXW4108 lo que trae conflicto.
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