Implement G.722.2 AMR-WB and EVS codec as FreeSWITCH module. Not a passthrough module, must actually be capable of transcoding to L16 (or whatever is considered the most native for 16khz stream in FS). Purpose is obviously to support wideband audio to wireless phones when supported. It will be our responsibility to license with VoiceAge.
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
...man scans a barcode which is his id, in order for the pos to identify him, then he scans a number of barcodes on receipts 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
Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDrive App. Need check the requirements from Pipedrive. [login to view 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
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.
Need ANSI C Programmer with FreeSWITCH module programming experience. Also, experience with AF_UNIX named sockets. Module needs to buffer sound bytes of audio from A leg into memory, package and deliver to another process upon silence threshold being met, and continue until timeout or message received from socket. You could start with mod_dptools
Hello. We are Looking for Call Center Solution based on Freeswitch (Multitenant) (Not Vicidial or GoAutodial) along with Source Code. If anyone have ready, contact us and provide us demo. We will purchase instantly. For more information you can chat with us. Thank you
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 an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.
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?
Hi, i would like to cooperate with an expert on FreeSwitch configuration. Just to make it straight, I don't plan to use any web interface as a plugin extension like FusionPBX etc. I want configurations from the filesystem. Currently, I just have two questions. But the person that will help me on those items i plan to hire with a monthly consultation
...from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with other freeswitch, asterisk
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Hello, We would like to pass a single field from out PHP program to a lin...that will: 1. Take the number as an input value 2. Attempt to call that number provided 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 [login to view URL]
...Priority musbe be have to work you can use any good Engine like asterisk,FreeSWITCH,Kamailio,OpenSIPS,etc More info check Attach file here is all screen shoot i need a complete wholesale Softswitch with a web management panel If you can make it please bid i dont have time for experiment i have bad Experience on freelancer so i will not pay any small
I am looking for an expert programmer in asterisk that can set 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 184.108.40.206
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
...phone bridge (1.4) on Asterisk PBX Version 220.127.116.11. / Java(TM) SE Runtime Environment (build 1.7.0_80-b15), Java HotSpot(TM) 64-Bit Server VM (build 24.80-b11, mixed mode) / FreePBX. It's partially working, incoming and outgoing calls I can make from Zoho crm, with Softphone. But I have the following 2 problems: 1. Outgoing calls made from zoho crm
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
A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign
...I have a NLU API por chat and text message plataforms. Now I'd like to develop a Voice Application using freewsitch for Voip Telephony. First I wan't to discuss what is the best architecture (Hardware, SO, Freeswich Version, freeswitch modules, failover, high avaliability), Install this Environment in my CloudServers, and then create a custom lua script
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. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
Hi i am looking for help on Big Blue Button Server setup, customization and branding along with performance optimization Knowledge of following Technologies may be required: Ubuntu, Web RTC, nginx, red5, FreeSWITCH, tomcat7, redis, Meteor, turn-stun coturn turn server
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 : [login to view URL] regards
...small businesses. I have just a few clients. On occasion, I go out of town for a week or so, 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