...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
...the phone does not exist on the database (both mobile and phone partner fields) it should automatically open the add new customer of POS. The functionality is similar to this module: [log ind for at se URL] but inside POS environment 2) An interface where the delivery man scans a barcode which is his id, in
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 ind for at se 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.
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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Hello, We would like to pass a single field from out PHP program to a linux/asterisk script/process that will: 1. Take the number as an input value 2. Attempt to...for 1 second from Asterisk/Trunk 3. Catch the return code of the attempted call 4. Pass the return code back to the calling PHP program We are using Asterisk 13.19.1 and FreePBX [log ind for at se URL]
I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit call...all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 18.104.22.168
i have a freepbx system and need to configure one sip detail but not able to use that details like as sip trunk. if i put same details on xlite or IP phone its working fine. so its a simple task but i need some expert one for this.
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
Zoho's phonebridge plugin supports up to Asterisk 1.4 [log ind for at se URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially
Hey, guys, I have a simple project. I have a FreePBX server that I need to build an inbound Sip trunk to 2 separate carriers + build inbound routes for them. It should be a simple process. Please respond to the bid with "What up Dingo" at the beginning of your message so that i know you have read.
I am looking for a complete working installation for FreePBX. The developers must supply my developers with working endpoints and 5 configured phone lines for inbound and outbound calling. Endpoint API must be handed over to developers. Must configure system to send an API call to external service whenever an inbound call starts. When a call comes
...system ready for online payment 3) Integration with CRM. Technology expertise 1) The engineer must possess good knowledge of Asterisk and Kamailio system 2) The engineer must have an in-depth knowledge of SIP layers 3) The engineer must be able to understand and troubleshoot systems such as FreePBX and A2billing instantly 4) The engineer must be able
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. ...system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [log ind for at se URL] regards
...and I need someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my clients. I will on occasion need projects handled such as repartitioning, adding a logical volume etc... Trust is a key because I will
Hi, I need an asterisk guru to help me acheive my goal. When someone reach my voicemail, I would like my to get a notification on my pager. That mean, system would have to dial a number, wait a couple sec, and dial 123 then hang up. This need to happen everytime someone leave me a voicemail.
Hello, I am looking for an Asterisks expert to code us the following dial plan. We will write into PBX local MySQL table records with phone numbers the Asterisks shoul...signup at ABC press 1, else press 2 or hangup". if user press 1 update database record to status 1, else to 2. With your bid include your pass experience with Asterisks and freePBX.
Hello, I want someone who can install and configure the Asterisk with FreePBX in CentOS Server. I don't want any custom feature for now but all the features which FreePBX and Asterisk offer should be configured properly. Thanks