Kamailio freeswitch fusionpbx jobs
I require someone to aid with some freeswitch esl work. converting 15 macros from asterisk to perl esl for freeswitch, work require interactive online user work.
FreeSwitch expert is needed right away. Will be tested on the work
We need a VoIP system as server and client side. Server side s...Server side should created by open source VoIP systems like asterisk Pbx, FreeSWİTCH etc. The VoIP system should include speech recognition (ASR, STT) and text to speech (TTS). The essential usage of this project is when you want to communicate with a deaf person using voice call. The deaf person can send text , the system read and sent into the call as speech. You talks into the phone and at the other end the deaf user can see what you said in written form on phone screen. For client side , We need mobile app for receive text and voice during call. Speech recognition for Asteisk Sppech recognition for FreeSwitch
Hi osypets, I noticed your profile and i would like to work with first of i want ot share details about me i am voip (freeswitch,asterisk) i just want to know if you are ready to have a company were both of us can work together.
OVERVIEW: My lead engineer has run into a problem with our lead product. He was tasked to install TLSv1.2 on a freeswitch server and then to produce a wireshark trace showing that the server was using TLSv1.2 for SIP traffic. We configured freeswitch like this: ./configure --prefix="$TS" --with-soundsdir="/storage/sounds" CFLAGS="-I /usr/local/ssl/include" LDFLAGS="-L/usr/local/ssl/lib" TASK: I need a qualified SIP developer to see why our freeswitch SIP is using TLS 1.0 for SIP connections, even though v1.2 is being specified. KEY SKILLS: Expert in freeswitch and TLS; ------------------------------------------------------------------------------------------ The deliverable items are as follows. 1.)...
Need a help for the following project. Project: Configuring FreeSWITCH with SIP Users in MySQL [Mod XML_CURL] so that two sip users of SIP client Csimple android app can dynamically register to freeswitch and calls each other. Baseline: Skill request: Please apply for this if you have actually hands on experience
We are currently in the process of build a voip infrastructure that we can sell to our clients. Right now we are running on fusionpbx but are having issues with t38 and our faxes working like they should. So we are in the process of testing other systems to see if we can figure out the best way to do what we want to do.
Hi! I have a VoIP server and a web interface built using Ruby on Rails to allow users to sign up, manage accounts, pay and automatically add the new user to the VoiP server (running Kamailio). Today my system is already automated with Stripe Payments. I want to add the same automation we use with Stripe, but for PayPal subscriptions (7 day trial with a $2 one time fee when sign up). I also need to integrate Btcoin payments using Stripe: In this case we must allow users to choose the period (3 months, 6 months or 12 months) because it does not allow subscriptions when paying using Bitcoin. Attached is a picture showing my current checkout screen. We can separate the payment options by TABs and use FontAwesome icons + text for each payment method
We have an existing VoIP environment and we are looking for an expert developer with the following skills for some development and implementation work: Asterisk A2B Kamailio PHP Apache Linux / Centos The work will involve adding new Asterisk & A2B features and also changing the existing environment. This will be 3 to 4 weeks of full-time work starting immediately. We are looking for someone full time initially with the view for more work! Details of all work can be discussed. You need to be available now and have the experience and skills.
I set up kamailio and rtpengine behind NAT, and make DMZ for kamailio server. I add advertise pub ip for kamailio in configure file with listen= advertise pub ip and config rtpengine with --interface=localip!pubip. now the problem is when I using UA from local call the UA on internat, verything is ok, except the SDP of ACK to local UA, the connection info c= pub ip, this should be local ip of kamailio.
I set up kamailio and rtpengine behind NAT, and make DMZ for kamailio server. I add advertise pub ip for kamailio in configure file with listen= advertise pub ip and config rtpengine with --interface=localip!pubip. now the problem is when I using UA from local call the UA on internat, verything is ok, except the SDP of ACK to local UA, the connection info c= pub ip, this should be local ip of kamailio.
I set up kamailio and rtpengine behind NAT, and make DMZ for kamailio server. I add advertise pub ip for kamailio in configure file with listen= advertise pub ip and config rtpengine with --interface=localip!pubip. now the problem is when I using UA from local call the UA on internat, verything is ok, except the SDP of ACK to local UA, the connection info c= pub ip, this should be local ip of kamailio.
I would like to set a twilio SIP Trunk to work with freeSWITCH on AWS
we are looking for some expert to quickly design a front end website (sample will be provided) and integrate with open source a2b (asterisk calling card ) or freeswitch or any other linux based voip telephony. Need the work done asap. You need to have good knowledge of php / mysql and knowledge of asterisk/ a2b/ freeswitch
Configure calling card access number in ASTPP (Freeswitch+Billing)
Configure calling card access number in ASTPP (Freeswitch+Billing)
We are looking for a lua developer to work with FreeSWITCH & FusionPBX to create customization's that will be put back into the community. The project is here on github --> All changes will be commit to github and must be accepted by maintainer Budget below is per week. We are anticipating that we are going to want to work with a team that will be able to continue to help grow the project for a long term. Project based pricing and Hourly based pricing depends on how large the request is from our side.
We have an existing voip platform with Kamailio load balancing a pool of Asterisk servers, we would like to install a few modules on Kamailio such as snmpstats and pike. Please only apply if you have experience with such setups.
We need someone who can configure open source G723.1 codec on our freeswitch. We already done with system all other codecs are working fine only giving problem with g723.1 codec. Note: Passthrough mode is already working, we want transcoding of g723.
We are a growing technology company providing services to US hotels and are in need of a system admin to join our team. Part-time, on call, during US business hours 09:00am ET to 09:00pm ET preferred. anytime in this window is ok - we are looking to roun...our team. Part-time, on call, during US business hours 09:00am ET to 09:00pm ET preferred. anytime in this window is ok - we are looking to round out our team. The selected candidate will join the team, learn our system and processes and work certain hours, and certain - standby hours until we can covert the position to full time - which will be, hopefully, in a couple months. We use Freeswitch, AWS, cisco, Unifi (Wi-Fi APs).. etc. If interested please send qualifications and availability [Removed by Freelancer.com]...
voip system Installation by opensips, freeswitch and astpp
We need someone who can configure open source G723.1 codec on our freeswitch. We already done with system all other codecs are working fine only giving problem with g723.1 codec. Note: Passthrough mode is already working, we want transcoding of g723.
Need a PHP developer skilled enough to help adapt an application to PostgreSQL (originally written to work with MySQL). The purpose of this is to be able to make API calls to our FusionPBX system using the XML_CURL module of FreeSWITCH (mod_xml_curl) to retrieve SIP credentials that will be used in auto-provisioning of client devices. Attached is a zip file containing the application that needs to be adapted.
I want to create a Salesforce CTI connector which will integrate with FreeSwitch. I need someone to work on a project to implement the required functionality to FreeSwitch to be able to communicate with SalesForce Open CTI connector (which will be build by another party).
...number manager app. (We will provide the API) 5: We would like to know how to add phones/provision them. 6. Brand Kazoo with our name and logo. 7. Show us how to whitelabel Kazoo for our resellers. 8. Connect Kazoo to our sip trunk for all inbound and out bound calls. 9. Once done, we would like a little training on Kazoo. Please DO NOT bid on this project if you do not know Kazoo, freeswitch, api or anything listed above. We are looking to hire someone tonight to start this project, so please don't waste our time if you can't do it. When bidding, please be sure to include a Yes or No to everyone of the points above and include your background with Kazoo platform and why you feel you are qualified to complete this task. Again, we DO NOT have time to waste ...
We have installed Kazoo on a single server on AWS. We are looking for someone who can do the following. 1: Test and make sure Kazoo is setup correctly and working as it should. 2: Set FreeSwitch config file to work with AWS NAT. 3: Add 2 service plans. 4: Modify the Number Manager App to use API from our other system so that numbers can be managed from the app. (We will provide the API) 5: We would like to have an auto provisioning app for phones, so if you know of one please let us know so that we can discuss having you set it up for us. 6: We would like an app that can bill. We don't charge per minute, we only charge per SIP trunk and per Device, so this should be simple. We would like to set our price for each service and have the app bill the reseller account. The r...
Hello there, we want to develop
We have a Freeswitch/FusionPBX Installation and we are having problems with endpoints running behind NAT. We need someone who can troubleshoot and remotely configure our VoIP Server so that the client problems are solved.
We are looking FreeSwitch setup & configuration on our dedicated server to support below functionality 1 to 1 Audio & Video call Audio conference Call – 100 Video Conference calls – 20 Call Recording (Audio , Video) DTMF IVR all as per freeswitch support must provide support for 1 year. We pay support charges monthly fixed amount as per agreed.
This is a task regarding kamailio configuration for pbxlt
I am looking for someone who has successfully integrate Skype trunk onto Asterisk or freeswitch to do the following. calls to skype are 1 way only, no outbound trunk is necessary 1. an incoming call is routed to asterisk / freeswitch via DID 2. asterisk / freeswitch will route the call to designated extension (skype) ID 3. extension (skype channels) will be able to transfer locally to another extension (skype channels) Thanks
I am looking for someone who has successfully integrate Skype trunk onto Asterisk or freeswitch to do the following. calls to skype are 1 way only, no outbound trunk is necessary 1. an incoming call is routed to asterisk / freeswitch via DID 2. asterisk / freeswitch will route the call to designated extension (skype) ID 3. extension (skype channels) will be able to transfer locally to another extension (skype channels) Thanks
i need to configure freeswitch or kamailio or openSIPS as proxy ,please bid if you have any experience , my project cost is fix 25$.
Need this done: %3A+CDR+Files+Integration Also will need to configure an example of a INBOUND route (client) and an outgoing DIALPLAN to my Asterisk server.
Need this done: %3A+CDR+Files+Integration Also will need to configure an example of a INBOUND route (client) and an outgoing DIALPLAN to my Asterisk server.
Dear Freelancers, We want to test ASTPP to use it as softswitch, we need to be able to test all features and be able to send and receive calls, We want freeswitch to be installed on a separate machine for redundancy and high availability purposes , We want also the backend DB to be installed on a separate machine. We will arrange the access needed on all resources with the winning bidder. Thanks
We are looking for a Voip Pro that can setup a Freeswitch cluster using FusionPBX. Requirements: 1. Quick delivery - before Christmas. 2. SIP TLS + SRTP. 3. Centrally managed DB for both freeswitches 4. Presence enabled. 5. Documentation & knowledge transfer Attached file. (Please see SETUP A)
Fusion rebranding,PHP custmization,menu redesign,admin page customization ,agent portal and reports, billing.
We have a server setup with fusionpbx. Here is what we have left. We need to fix our email sending to send voicemail and faxes to extension emails. We need our fax extensions setup to receive faxes correctly. We need to setup outbound routes to be able to send faxes from the web interfaces. We need a different voice prompt setup for outbound call confirm. We need our DISA to work properly. Setup codecs to get the best quality on the gateways and best voice quality on calls. We also need to look into High availability setup.
I have a Freeswitch with FusionPBX. When I send the call to one of my vendors in G729, the audio is really bad, metallic and robotlike but in G711 the audio is perfect. I set the outbound codec to PCMU or PCMA (G711) but since the incoming call is in G729, the Freeswitch always offers the F729 as first choice and the vendor takes it. I need to send the outbound call to the vendor offering ONLY PCMU or PCMA, without G729.
We are a new and successful startup. In order to grow up more, we need to develop a custom telephony solution based on Asterisk or Freeswitch. This solution must be interfaced with our CRM written in PHP / Laravel.
We are a new and successful startup. In order to grow up more, we need to develop a custom telephony solution based on Asterisk or Freeswitch. This solution must be interfaced with our CRM written in PHP / Laravel.
Hello, I've look your perfil and see the Autodiler solution based on Freeswitch. I need some solution for integration with FusionPBX you can made it with you developed solution? Your autodial solution is based on Goautodial? Please chat me on skype: for more information. Best Regards, Rafael
Responsibilities: - Developing new SIP based components / features for server - E...environment Strongly Desired: - Familiar with all modern software libraries available for the Linux platform - STL, boost - Expert level know-how in fast/light-weight string parsing algorithms - Expert in application debugging / profiling tools (Valgrind, Google Perftools) Nice to have: - Previous experience in open source projects - Agile (SCRUM) development process knowledge - Hands on FreeSWITCH, reSIProcate, sipX, SER, JAIN SIP, Kamilio - Actively participating in the creation of RFCs / keeping yourself up to date with the current drafts - Hands on working with cfengine framework - RDBMS design principles and concepts (PostgreSQL) , good SQL knowledge. NOSQL database concepts...
Dear Freelancers, We are looking for someone who can build optimization in voip traffic. Please find the below key points. 1. We have Freeswitch and softphone clients in android and IOS platform. Android Client source is csipsimple and IOS is Siphon. we need make optimization between switch and client. Actually we using G729 codec in switch and it is consuming around 48 Kbps for each call Because of overhead. Please refer a product that currently available in voip business " Itel Byte Saver" you will get more idea after reading about itel byte saver' We need slimier like that or same :) We need implement below technique to get better result. 1. RTP Multiplexing and Delta- Multiplexing 2. Voice / Packet Header Compression 3. Packet Header Reduct...
We are in need of a Fusion PBX guru to help with an existing hosted pbx on AWS. Requirements are to, Update to latest stable version of Fusion PBX Check and fine tune existing installation for multi tenancy environment on AWS EC2 Configure call flow for multiple clients including IVR, Fax to eMail, eMail to Fax, Ring groups for departments...PBX Check and fine tune existing installation for multi tenancy environment on AWS EC2 Configure call flow for multiple clients including IVR, Fax to eMail, eMail to Fax, Ring groups for departments, Ring groups for users with multiple devices, Extensions, Voicemail to eMail automated reporting A critical requirement of the expertise of the following is a MUST! Linux AWS EC2 Fusion PBX Freeswitch Grandstream Snom Yealink Poly...
Hi there, I am looking for help in a freeswitch projet. We want to setup an E1 line with a 1TE133F digium card. This is an humanitarian project for the Ebola response in Africa
This module will include two parts: 1. Setting up a dialing server 2. Writing a php class for handling operations with the server. More details in the attached PDR. Please submit your suggestion as accurate as possible for price and time frame.
