Kamailio freeswitch fusionpbx jobs

Need someone to: 1. setup Kamailio (openser) sip server. 2. Configure to disable authentication for all requests 3. an API to sign-up new sip account

We need a professional for setting up and configuring an freeswitch. We want a freeswitch server which can handle up to 700 (current calls) incoming media calls (inbound, incoming calls) and for this we need someone who can configure for us the freeswitch server. The ivr of the freeswitch needs to be configure for this issue as well

We need a professional for setting up and configuring an freeswitch. We want a freeswitch server which can handle up to 700 (current calls) incoming media calls (inbound, incoming calls) and for this we need someone who can configure for us the freeswitch server. The ivr of the freeswitch needs to be configure for this issue as well

install asterisk / freeswitch; for billing install a2billing and need to modification and customization

Freeswitch to Asterisk will be pending and update later

Vbilling is a billing software for the wholesale trades that uses FreeSWITCH. We provide a 7 Debian VPS with SSH access The installation script is on the site but requires adaptation for Vbilling work properly.

We require ASTPP customization. - Resellers need to be able to create their own invoice and warning templates. These templates are stored in the database. Currently ASTPP uses global templates. - We'll need to be able to create openSIPs/FreesSWITCH profile aliases so that we don't need to creat...be able to see see origination rates. These rates will need to reflect the rates that the reseller or administrator specifies. For example an administrator will import rates and add a 5% markup for resellers. The origination rates with a 5% increase will need to be displayed when the reseller views their origination rates. The same will apply to clients. Note that OpenSIPS is used as a FreeSWITCH proxy and there is a dedicated database server. All the servers are part ...

I need the Astpp () billing software installed and running over "Freeswitch" () running on a server on Centos 5.5 I try but dont have the time to continue this project. Will be done remotely on a hosted server. If successful may work in further projects Any further inquire please pm me.

We have a Sipxecs IP-PBX that is working with an Audiocodes Mediant 1000, digital gateway. We need another IP-PBX (prefer FreeSwitch) to work with mediant 1000 for inbound and outbound call handling. Of course both IP-PBX's users should be able to communicate. This task is basically needs configuration skills. Will provide remote access to servers. More detailed information will be provided before starting the project.

Part 1: We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We will require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We will provide remote control access to the server with the SIP line. (You are required to pick the server parts and provide a justification for the choice of parts. Please keep in mind we are hoping to scale the service quickly and the system should be able to accommodate multiple servers.) Once the system is built, we need support for a limited time and training of our technical ...

Part 1: We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We will require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We will provide remote control access to the server with the SIP line. (You are required to pick the server parts and provide a justification for the choice of parts. Please keep in mind we are hoping to scale the service quickly and the system should be able to accommodate multiple servers.) Once the system is built, we need support for a limited time and training of our technical ...

I need support configuring a Cisco 7970 to work on a PBX hosted on PfSense. The package installed is Freeswitch, but asterisk can be used too. The phone has been already upgraded with a sip firmwire, and the phone has been configured with a SEP Conf xnl file. At present it is unable to register with the package hosted on the PF Sense box I am very busy and could do with some help getting this to work. Its a configuration issue and need someone to talk me through the process of getting the equipment to talk to each other. On completion I will need to be clear on how i configure a System by myself. I can work with remote access and Skype to complete this task.

We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. Thomas@ SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

We have a Programmer from Bangladesh disappear. We had operating system but now it is broken :( Need it running for a demo and minor changes. Thomas@ SKYPE: [The administrator removed this message for containing contact details which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

We currently are using Newfies / Freeswitch system We would like to have campaigns automatically retrieved from FTP and configured in Newfies so we don't have to do it manually. We FTP the file to a folder, system will cron to grab every 5-8 minutes. The zip will contain 2 wavs (live and answer machine versions), a csv list and , stamped with the projectID number. The list csv will be returned as (projectID)(zip). The call records will populate in the additional columns in this list before it is returned to FTP. We are NOT worried about any frontend work, as this automation will not require any frontend use. Ready to start this now. I'm online frequently

We need a Freeswitch based system which can handle basic IVR and outbound calling and dialer functionality thru a SIP line. The system should be capable of handling high volume of calls. We would require your assistance with installing, configuring and building an outbound call bridging system to provide add based calling to our users. The system will also have a basic IVR system. We would provide remote control access to the server with the SIP line. Once the system is built, we need support for a limited time and training of our technical staff on the built system. We are open to both hourly billing or fixed price models. Part 2: We would also require an advertiser portal where the user can log in and add advertisements to the phone call.

Setup of a DebianWindows server with Freeswitch and GsmOpen. I just need the basic assistance to setup the system.

Hello We already have FreeSwitch Setup with Open Source VOIP billing Solution. But we need an expert on FreeSwitch with Voip billing Solution who can tune up the configurations and make our dedicated system to run high capacity wholesale voip business. We want following tasks to be done. 1. All FreeSwitch Configuration and Tunning. 2. VOIP billing features and billing must be working properly. 3. A very detail step by step and easy to understand Manual for features. It should be made in such a way that any non technical person should be able to understand it and start using billing solution instantly. It should be separately made for Admin, Sub Admin, Reseller, Sub Reseller, Users, Calling Card, Call Shop Users, Providers. 4. Complete System

Hello, We require the building of a VoIP Switc...to the relevant SIP Account the DID is allocated to. We need to allocate a cost on each channel so each call has a cost to it based on prefix. At month end, we must be able to draw out a report stating the inbound calls via the relevant channels and its cost, with a total of it. The CORE should be something sturdy, with data saved in a database. Opensource CORE is welcome. Either Kamailio, A2B, FreeSwitch, Kazoo, or any other robust switch and billing system. There is NO outbound VoIP in this switch, its purely inbound. Ultimately, we need to know which provider is calling our DIDs, and we need to allocate a cost so we can bill them for sending calls to us. We need someone with good experience to handle this proj...

I am looking for freelance who can help me to implement SIP 2 RTMP gateway with help of freeswitch. I have tried to implement it but facing some issues. I got success till: enabling mod_rtmp connecting to rtmp server with flash client provided with freeswitch source code login using username and password successfull call to 888at failure at: trying to make calls to extensions

Hello, We require the building of a VoIP Switc...to the relevant SIP Account the DID is allocated to. We need to allocate a cost on each channel so each call has a cost to it based on prefix. At month end, we must be able to draw out a report stating the inbound calls via the relevant channels and its cost, with a total of it. The CORE should be something sturdy, with data saved in a database. Opensource CORE is welcome. Either Kamailio, A2B, FreeSwitch, Kazoo, or any other robust switch and billing system. There is NO outbound VoIP in this switch, its purely inbound. Ultimately, we need to know which provider is calling our DIDs, and we need to allocate a cost so we can bill them for sending calls to us. We need someone with good experience to handle this proj...

Configure Kamailio to make sure eveyclient can chat with each other and be called with each other. Kamailio 4.1 is already installed, but the configuration needs to be fixed. When we install the system, all of the user can login the system. However, some UA behind the NAT can't get the message. for example. When A ---NAT--> sip server -->NAT-> B (SIMPLE CHAT message) I have already make sure A. B enable the STUN. However, SIP sever will send the chat message to "B's private IP ()" I just want to make sure , in all of the conditions, A and B can send the chat message with each other. Can someone help us to set the configuration ?

I am looking for someone to assist in setting up load balancing using OpenSIPs, and Asterisk. FreeSwitch may also be used if it is needed. I need a load balancer and SIP proxy's and media server. The entire setup needs to be Highly Available, so that any single server can go down without any services failing permanently. I need written proposals of experience, and how many servers you would need to make this setup work. The end goal is a Multi-Tenant hosted PBX system.

Install, configure and document billing system Document configuration of billing system 1 hour walk through of billing system Secure server (firewall/fail2ban)

Expert needed in VOIP, SIP, Kamailio, Amazon EC2..

PLEASE DO NOT BID IF YOU HAVE NEVER DONE THIS BEFORE!! I need this implemented on Freeswitch or if you can recommend something that will be able to do about 250 calls. 1)Integrate radius with jerasoft for radius auth/billing 2)Accept inbound DID and send back IVR announcements for calling card 3)make sure correct ANI is being sent to jera , so we can do ANI (pinless) calling. 4)allow PIN entry (retry 3 times, if incorrect pin is entered) 5)allow destination number (retry 3 times) 6)Prompts in English when call arrives from DID1 and ability to give different language prompts when call arrives from different DID's . 7) if customer calls and his balance is depleted , then system must give option to enter refill voucher 8) If incorrect destinat...

My company is seeking ones Knowledge of Vicidial , MySQL, Ajax, Php & Bootstrap. We are running 1 large Database and 11 Agent servers. Realtime Advanced Stats to Display on a client admin panel with login prompts. Must have a proven track record and well knowledge of the exact tasks needing performed. Must have knowledge of Freeswitch on a large cluster as well. We send all traffic to our own switch and you will need to be able to retrieve the db queries from astpp db's too.

I am in the process to configure Kamailio with Asterisk for personal use. Below are the setup details: - Asterisk and Kamailio on one VPS with 1 public IP - Asterisk listening at Port XXXX - Kamailio is listening at port YYYY - No P2P calling Kamailio is acting as registrar and Location server at the moment. INVITES are being forwarded to asterisk by Kamailio. I AM FACING THE NAT ISSUES BECAUSE THE EXTENSIONS ARE BEHIND NAT. ONE WAY AUDIO OR NO AUDIO AT ALL WHEN CALLING. Tried rtpproxy but it seems I am not the expert of configuring routes in Kamailio so I am looking for someone who can help me to fix and understand the issues. Budget: Setup is for my personal use so I don't have any budget in my mind but if you can tell me how much it w...

I am in the process to configure Kamailio with Asterisk for personal use. Below are the setup details: - Asterisk and Kamailio on one VPS with 1 public IP - Asterisk listening at Port XXXX - Kamailio is listening at port YYYY - No P2P calling Kamailio is acting as registrar and Location server at the moment. INVITES are being forwarded to asterisk by Kamailio. I AM FACING THE NAT ISSUES BECAUSE THE EXTENSIONS ARE BEHIND NAT. ONE WAY AUDIO OR NO AUDIO AT ALL WHEN CALLING. Tried rtpproxy but it seems I am not the expert of configuring routes in Kamailio so I am looking for someone who can help me to fix and understand the issues. Budget: Setup is for my personal use so I don't have any budget in my mind but if you can tell me how much it w...

Expert needed in VOIP, SIP, Kamailio, Amazon EC2..

...would also be useful to others. I'm surprised there isn't a site that already does this. I don't want to deal with applicants that haven't done this kind of work before. If you are going to pursue this job, please make your relevant experience clear. Also, I am hoping I can find someone that can easily accomplish far more than this simple introductory job -- someone with experience with freeswitch or equivalent, and that has experience in building call centers. Here's what I'm looking for for this simple introductory job: In the testing website you produce, the user would be able to go to the test site, and choose two things: 1) which of the 12 api speech recognition solutions they want to test, and 2) produce a recording in one of 2 w...

TO ONLY PEOPLE WITH EXPERIENCE WITH DONGLES : i need to setup a Voip server to use as gsm gateway for outbound calls using huawei dongles. using chan dongle for asterisk, gsm open for freeswitch, or a private module. fully manageable with web GUI with the following feature: -ability to have full control over trunks/extensions : time schedule of calls, time gap between calls, limit unique dials, maximum duration per day/hour … -ability to send call of a previously called number to the same exentsion used for the previous call, then to next dongle. -see credit per dongle based on ussd and pdu msgs and parse them correctly on the table. -send credit to one or group of dongle on gui using ussd sms and calls. -automatically detect blocked cards of dongles and disable them...

I need some one that can debug a attended transfer tonight..I'm looking for someone that knows freeswitch and can support me ongoing..

...would also be useful to others. I'm surprised there isn't a site that already does this. I don't want to deal with applicants that haven't done this kind of work before. If you are going to pursue this job, please make your relevant experience clear. Also, I am hoping I can find someone that can easily accomplish far more than this simple introductory job -- someone with experience with freeswitch or equivalent, and that has experience in building call centers. Here's what I'm looking for for this simple introductory job: In the testing website you produce, the user would be able to go to the test site, and choose two things: 1) which of the 12 api speech recognition solutions they want to test, and 2) produce a recording in one of 2 w...

I purchased this from Anveodirect: - 2 DID #s - 1 trunk/channel I did this so that if anyone calls either of the two DIDs it will share the trunk. In case people call both DIDs at the same time, only one call will go through due to having only 1 channel. With Anveodirect, I think I am supposed to have my own PBX, but I do not want to host it. I am looking for a cloud based PB...host it. I am looking for a cloud based PBX where I can set everything up. I currently have , and it is free, but I have not idea how to set it up to get it to work. I also need to add a voicemail to each of these #s and to have it send any voicemail as an email to me. I am hiring someone to train me on how to set this up. Thank you. Keywords (ignore): Asterisk, Freepbx, freeswitch...

Hello All, I need someone who is very familiar with freeswitch that can do the following task configure multi-tenant setup create dial plans configure all features as needed. Please respond to this with "Hello Dingo" So that I know you have read completely.

I want to learn how to use OpenSips for integration with freeswitch or asterisk. I know SIP, linux and VOIP well, but need to understand how to setup, configure and secure a carrier grade voip network. If you have extensive experience using SIP proxies,RTP proxies, etc.., Then provide a brief synopsis of your experience and what you'll provide training on.

Hello, I need freeswitch installed with the following features enabled. The latest version of free switch Multi tenant Domain routing VoIP extensions-with account codes Voice-mail Inbound calling outbound calling-- with account codes Ring groups Inbound queues auto attendant inbound fax outbound fax Conference Bridge I would like the features to be enabled in the order listed above. I will need sample files for each features as well as brief instructions on where files will be located for each feature. I will provide dedicated system Debian 7 (Wheezy) I will escrow amount for each feature and will release one by one until all features are completed. Please provide total amount to be charged for each feature. Please let me know if you have any questions

We currently are using Newfies / Freeswitch system We would like to have campaigns automatically retrieved from FTP and configured in Newfies so we don't have to do it manually. We FTP the file to a folder, system will cron to grab every 5-8 minutes. The zip will contain 2 wavs (live and answer machine versions), a csv list and , stamped with the projectID number. The list csv will be returned as (projectID)(zip). The call records will populate in the additional columns in this list before it is returned to FTP. We are NOT worried about any frontend work, as this automation will not require any frontend use. Ready to start this now. I'm online frequently

Hello I need what you configure astpp with freeswitch to work how we need. Please contact me. for more details... Thanks

We need what you configure the ivr, to work how we need. You Edit and configure the database to work with Prepaid card. Some more information, contact. Time to work is the usa time.

Hello I need what you configure astpp with freeswitch to work how we need. Please contact me. for more details... Thanks

I need an install of ASTPP with Freeswitch in an Openvz container. The OS can be either Debian 7, Ubuntu-12.04 or CentOS-6.5. What is your recommendation? This can be a single server install as it is a proof of concept. If my demo is successful, I will need help scaling to separate DB, Opensips and Freeswitch servers. I will run wholesale traffic through this switch(mostly China CC). For this single install, would you recommend installing Opensips? I will need all functionality working (including CDR reporting). I will need some documentation on how to configure correctly. All provider and customers will be authenticated by IP and Tech prefix. I offer each Customer a Standard, Silver and Premium route based on tech prefix. I blend my providers to achieve this. I w...

I need an install of ASTPP with Freeswitch in an Openvz container. The OS can be either Debian 7, Ubuntu-12.04 or CentOS-6.5. What is your recommendation? This can be a single server install as it is a proof of concept. If my demo is successful, I will need help scaling to separate DB, Opensips and Freeswitch servers. I will run wholesale traffic through this switch(mostly China CC). For this single install, would you recommend installing Opensips? I will need all functionality working (including CDR reporting). I will need some documentation on how to configure correctly. All provider and customers will be authenticated by IP and Tech prefix. I offer each Customer a Standard, Silver and Premium route based on tech prefix. I blend my providers to achieve this. I w...

Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www

Have a look at my project regarding VOIP. Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones (http://www

Please find requirement below: Freeswitch 1.2.23 Configured as Media Server - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones () - Voicemail to Email - SIP presence configured (alternatively configured on Kamailio) - SIP Instant Messaging (alternatively configured on Kamailio) Kamailio 4.1 Configured as SIP Frontend Server in front of Freeswitch - Multitenant Configuration - SRTP and TLS configuration - Configuration for the following clients - Counterpath Bria () - Global IP Juggler () - Snom Hardware Telephones

We currently are using Newfies / Freeswitch system We would like to have campaigns automatically retrieved from FTP and configured in Newfies so we don't have to do it manually. We FTP the file to a folder, system will cron to grab every 5-8 minutes. The zip will contain 2 wavs (live and answer machine versions), a csv list and , stamped with the projectID number. The list csv will be returned as (projectID)(zip). The call records will populate in the additional columns in this list before it is returned to FTP. We are NOT worried about any frontend work, as this automation will not require any frontend use. Ready to start this now. I'm online frequently

We are a small IT-service company which provides besides general IT-services specific products. These products are build on linux and need developement and maintenance. One of this products is a virtual multitenant PBX based on freeswitch with fusionPBX, everything completely opensource. Your job would be to maintain the PBX, be in contact with the developers of the software and client-activation. Furthermore would you manage another product based on "PYDIO" and general assistance in our Linux environment. Knowledge in Microsoft environment is appreciated, since third party software connected to the PBX is running on windows server. Strong understanding of general networking (routing, switching,...) and virtualization is absolutely required. Right now we a...