Freeswitch dialplan Jobs

NEEDED - Expert at Freeswitch (preferable) and/or Asterisk. PROJECT - Mimic the development of - we will need both the web interface as well as the voice conference functionality. Use the following credentials: User login: ttest8520... Pass: abcd1234 I couldn't type in the "at" term in the username above, so please replace "..." with the "at" symbol. The above should not be misconstrued as contact information as the above email is a generic email I do not use. Thank you, PB.

NEEDED - Expert at Freeswitch (preferable) and/or Asterisk. PROJECT - Mimic the development of - we will need both the web interface as well as the voice conference functionality. Use the following credentials: User login: ttest8520... Pass: abcd1234 I couldn't type in the "at" term in the username above, so please replace "..." with the "at" symbol. The above should not be misconstrued as contact information as the above email is a generic email I do not use. Thank you.

NEEDED - Expert at Freeswitch (preferable) and/or Asterisk. PROJECT - Mimic the development of - we will need both the web interface as well as the voice conference functionality. Use the following credentials: User login: Pass: abcd1234

I am looking for an individual with experience with Freeswitch/Asterisk implementations. The winner will be able to develop a full featured web interface for provisioning new customers and other resellers using the Freeswitch platform. The part that will be tricky is making this a sellable product to other VoIP resellers and provide a turnkey hosted solution that includes billing. If I am impressed with your work, I will call on you again to perform additional tasks. We are bundling VoIP with DSL, Satellite TV and Cell phone plans. This could be an opportunity to become a full time employee for the right person. The hardest part is going to be making Freeswitch work properly with your custom GUI and billing. If you think you have the skill, bid away!

...IMPORTANT: The system must be scalable so when the capacity is reached on the fist server, a second..third can be added. You will provide me with all the required configuration on how to add an extra telephony server when the current one has reached capacity. **You are required to utilise:** -**A good SIP proxy to take care NAT bursting and media RTP issues** (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reasons behind your selection.) **-A2billing for the billing part and web interface part.** Here you are required to enable the following common functions: 1)Voicemail: Users should be able to see (and listen) to their voicemails via their A2Billing Customer interface. Voicemail facility is not going to be charged to the use...

...first server will play the role of a sip proxy, load balancer, SBC and media proxy. The other two will be telephony servers (i.e asterisk) that will handle the calls. The system must be scalable so when the capacity is reached on the fist two servers, a third can be added. You are required to utilise: -A good SIP proxy to take care NAT bursting and media RTP issues (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reason behind.) -A2billing for the billing part. Here you are required to enable the following common functions: 1)Voicemail: Users should be able to see (and listen) to their voicemails via their A2Billing Customer interface. 2)Follow me number. Users should be able to select a follow-me number that will be billed according to ...

...first server will play the role of a sip proxy, load balancer, SBC and media proxy. The other two will be telephony servers (i.e asterisk) that will handle the calls. The system must be scalable so when the capacity is reached on the fist two servers, a third can be added. You are required to utilise: -A good SIP proxy to take care NAT bursting and media RTP issues (OpenSips?Kamailio?Freeswitch? You need to select which one and justify the reason behind.) -A2billing for the billing part. Here you are required to enable the following common functions: 1)Voicemail: Users should be able to see (and listen) to their voicemails via their A2Billing Customer interface. 2)Follow me number. Users should be able to select a follow-me number that will be billed according to ...

Seeking someone with experience using Freeswitch. We are running an IVR and Freeswitch keeps crashing. We are also using LUA and SOX for audio conversions. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature of the deliverables): a) For web sites or other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends to distribute: A software installation package that will install the ...

Hi, I wish to setup a asterisk (AsteriskNow, Freepbx, tribox etc) on a virtual server. Once this is completed i will probably need to create a dialplan which links to a mysql database based on user/call input.

I need freeswitch installed on Ubuntu remote server along with freePBX. I will provide ssh access to the server. This could result in an ongoing position for the person/company that gets the wins.

Hi, I want the following implemented in my dial plan: 1) All extensions are of the form 278XXXXXXX 2) My users must be able to dial 08XXXXXXX and dial the corresponding extension (i.e.: replace 08 by 278 and dial the resulting extensions) THIS IS FOR TRIX BOX! MAKE SURE THE CONFIG DDOESN'T GET OVERWRITTEN AFTER I USE THE GUI!!! NEED SOMEONE TO DO THIS WITHIN THE NEXT FEW HOURS.

...to do it, but recording on the server would be a great advantage. If this is too difficult, let me know and maybe I could drop this requirement. **OUTGOING RULES:** * Local extensions dial each other in 3 digit format, like 201, 202, 203 etc ? * Outgoing calls are dialed directly without any prefix ( without 9 for example ) ? * Dialplans are set up to dial immediately when dialplan rule is matched without having to press DIAL or OK or "#" after phone number. We have 10 digit dialing here and two local area codes. I will provide more detailed rules for the dial plans to be used. There are just a few of them, and they are pretty simple. ? * Special calls go through landlines only, like 911, 811 etc. (I will provide the rules) ? * So...

...include long distance charged to the defined countries. The dial plans DID’s defining the local calling areas: Each City, roughly 30, needs to be configured so customer calls in the defined are dialed as a local call. Customer's local calling area is defined by which calls the customer needs to dial 1 before the number. We can (and have) defined the local calling areas as contexts in the dialplan. For example, a caller in Location A can call other numbers in and around Location A by dialling the area code and exchange (first 6 numbers) followed by the local (last 4 numbers). If that person wants to call outside of the Location A area they are forced to dial a 1 before the number, or the call will not connect. When the billing is done, it only looks for calls wi...

We're having an issue with our FreeSWITCH installation. With dahdi / oslec installed we get no dial tone detected when trying to make an outbound call. We're using a Sangoma AFT A200 FXO card. We can provide an SSH remote login.

I need someone to write up a script that will do the following (All of the script must be in php and sql for asterisk) caller calls -> system looks up phone number in sql and updates all the caller info then looks up user and ip address -> system pings the ip address and responds to caller if the ping goes thru or not, then looks if the user is registered or not. (more info upon choosing a programer)

Current Situation: We have some Sip Clients registering on the asterisk box and if they dial a number the call will be forwarded to another Switch where the call will be sent to a carrier. What we are searching for: If the users dials a number which is registered as cli from another user on the asterisk box the call should go directly only via the asterisk box to the other sip client. If the calls will be connected only using the asterisk box there should be voicefile telling it to the calling party. -) There should be a cdr list including only calls which have been connected. -) No call rating is required. -) What happens if one or both user are behind NAT? -) What happens if the called party is busy?

Need Freeswitch GUI installed - proposal for customer self-registration and customer self maintenance as well as call routing and fail-over call implementation - please send email if more info needed, Also need Ruby on Rails tech support for website that needs a new "gem" installed. Project can become part-time permanent support project for the right candidate. Thanks Chris.

Seeking someone to install freeswitch on a remote server. SSH access will be provided.

We are locking freeradius billing system with admin, multilevel agent, customer & signup, calling card, call shop, wholesale module. Billing GUI must design in php or perl language & database in mysql. Actually, I m planning to start 1 server with freeswitch + freeradius billing& second with openser + freeradius billing. Developer has to provide documentation and source code for system.

Good systems administrator for Asterisk, Openser, FreeSwitch To provide support and installation help on a monthly contract basis. Configure new or Modify existing configurations, etc. Good knowledge of MySQL, Asterisk realtime, Openser/Kamailo routing, Work without supervision. A proven candidate may have 1 year worth of constant work. Good GAF rating, experience and record necessary.

We need someone who can both configure FreeSwitch and also make some changes to the FreeSwitch source code. We prefer to have FreeSwitch running on a Windows server.

...have astmanproxy on windows to freeswitch which is an asterisk derivative. I planned to first port astmanproxy to windows supporting asterisk and after that add freeswitch support using Poco... But meanwhile I've found a cross platform C library called esl which is already compiles on windows and linux too and supports freeswitch.... It's a console application which connects to freeswitch and listens for evenets and can can send actions to it... But it lacks some functionality that astmanproxy already have: -ability to run as a deamon on linux or as a service on windows. -handle lost connections to freeswitch and add automatic reconnection interval -start a listener thread for client connections and forward events and a...

Create an outbound dialing application using freeswitch and MySQL. Application will be a modified voice broadcast system. Dialer must handle many simultaneous calls. Complete details of project will be sent to applicants only. Please message to see full details. ## Deliverables Please contact via private message to view all project details

1. Provide sample MySQL records to us so we can perform field mapping and import testing. 2. Allow us to connect remote to your TEST FreeSWITCH so we can perform additional testing.

We develop and distribute a VoIP billing application (<>) . Right now we need assistance integrating with FreeSWITCH for record polling/retrieval. We need someone to assist in the following tasks. 1. Provide sample MySQL records to us so we can perform field mapping and import testing. 2. Allow us to connect remote to your FreeSWITCH so we can perform additional testing. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows (depending on the nature of the deliverables): a) For web sites or other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables

We are running a linux based PBX server (i.e. asterisk / freeswitch) and need to change SIP providers for termination. I need this done immediately. Once you are awarded the project, we can connect on Instant Messenger or Skype. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends...

I’m looking for a Freeswitch (VOIP) engineer/programmer to help with the following: •Multitenancy - Enterprise/Carrier configuration (to host more than 1 company phones, IVR, Voice Mail and Billing) •MWI Support •NAT Support (for remote phone registration) •Conference Server (per hosted company) •Voicemail Server (per hosted company) •SBC (Session Border Controller) •PBX (per hosted company) These are some of the features I would like to start with and add more features later. I’m looking for a long term partnership to help build this for our company.

I need someone to setup up one server with Freeswitch or Sipx. Some lines will use skype, some will use other providers. The server has 4 purposes, 1. company pbx 2. for a teledicatation system (almost like a voicemail server that is able to record multiple messages with one call). 3. And a voice broadcast system ( a system that calls out numbers from a database with custom messages. 4. Fax server In all examples I would like Freeswitch to use its own voice for the prompts and messages. Using the TTS engine ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web site...

I need someone to restart Freeswitch server. Also need a script that I can run to restart freeswitch in the event it stops again. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends to distribute: A software? installation package that will install the software in ready-to-...

We have an existing Asterisk system in place, we're looking for someone to make some changes to it. Specifically: ...extension which will play a recorded message telling them how to find it. Once they enter in the customer ID #, asterisk should then execute a command and pass that ID # as an argument. The command will then return a '1' or a '0'. If a 1 is returned the caller should be passed along to an already created phone queue, if not they'll be sent to listen to another prerecorded message. *Modify dialplan so that if all members of a queue are on the phone it places caller in a holding queue and somehow alerts call center staff that a call is waiting (This is an open point that we want feedback on and its exact requirements may change, yo...

Summary: 1. It is expected that this app will involve PHP for a web-based solution, plus SQL (Express preferred) and Freeswitch (open source telephony). This is a phone-based solution which shall be accessed and configured by a web-based application. 2. The app is designed to accept calls from Callers (who report to Supervisors) who will call a toll-free number (plan is to use a SIP service, so app must not involve any special telephony hardware) and leave messages which the app will process in a variety of ways. App must work in Canada, and will eventually reside on a hosted sever, but for development and testing purposes, it will be acceptable to write code that can be executed from installation on a local machine (intranet). ## Deliverables 1. PHP (or equivalent)-based app. wi...

...only, will consider options). 1. PHP (or equivalent)-based app. with screens to enter Subjects, Supervisors, and Reports. Will consider programming languages other than PHP. 2. Freeswitch scripting or other open source that will work with a SIP provider 3. MS SQL (Express) 4. Software solution only, I do not want any special telephony hardware – needs to work from a desktop/server 5. XP/Vista 6. Open source Test To Speech engine. Open source approach preferred to control costs. Summary: 1. It is expected that this app will involve PHP for a web-based solution, plus SQL (Express preferred) and Freeswitch (open source telephony). This is a phone-based solution which shall be accessed and configured by a web-based application. 2. The app is designed to a...

...high a cost, please describe how your proposed solution will manage multiple outbound calls scheduled for the same time. Detailed Requirements: 1. The preference is that the app will contain a built-in recording function to record wav files to be used by the app, and this function must record wav files at the appropriate Hz, Bit, Mono parameters to work with the open-source telephony app (e.g. Freeswitch). If the app cannot contain this function (or if too expensive) then at minimum the app will allow the Admin to easily organize, store and retrieve wav files that have been recorded. In the description that follows, it is assumed for simplicity that the recording function can be built-in. 2. The app will require Admin and Operator roles. The app will allow data entry for Ad...

Seeking someone with experience building dialplans using freeswitch software as well as managing and setting it up. Must have experience with javascript, php and mysql. We need some help building dialplans for various projects. Please tell me your experience with freewitch and send me some examples of your work if possible. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer...

Looking for asterisk/linux experts to provide a elastix customization. We need a usb installation based software where we can deploy images to new computers. The image deployed has to contain a recovery partition where the customer can restore the PBX server to original status and than upload the dialplan . extensions from a previously saved backup.? The customization will include a all-in-one appliance where the box can be used also as a router with integrated prioritization for voip . ## Deliverables 1 customized boot image? 2. deployment to new server via usb stick or iso image , cd? 3 contain recovery partition where customer can restore original status and eventually restore previously backed up extensions dial up plan? 4 shall include all in one appliance with routi...

...start up company and looking for a OpenSIPs/FreeSWITCH expert to help in setup proxy server. We are using CentOS and would like to use SerMyAdmin web tool unless you have better GUI based other recommendations to do the same. We have multiple clients application (SIP UA) which I want them to register to OpenSIPS proxy server and have this proxy server to talk to FreeSWITCH which will manage SIP trunks for outbound calling. I would like have capability to register and maintain multiple SIP trunks on FreeSWITCH and use them based on predefined dialplan/rules. We must be able to setup and distinguish billing details details and CDR etc.. and all CDRs should be handled by OpenSIPS server. Please bid only if you have experience with OpenSIPs/FreeSWITCH in p...

I need someone who understands C code and has experience with audio. I am using a PBX system called Freeswitch and need to detect beeps from various answering machine. The code works for many carriers but we need to enhance it to support a few others i.e. "Sprint"... for some reason, its not detecting the beep on some carriers. I'm sure its a threshold setting or algorithm we need. Here is the script we are using: Here are some additional threads:

...would like to use SerMyAdmin web tool unless you have better GUI based other recommendations to do the same. We have multiple clients application (SIP UA) which I want them to register to OpenSIPS proxy server and have this proxy server to manage SIP trunks for outbound calling. I would like have capability to register and maintain multiple SIP trunks on proxy and use them based on predefined dialplan/rules. We must be able to setup and distinguish billing details details and CDR etc.. Please bid only if you have experience with OpenSER/OpenSIPs in past and confident to do the job. This is small bid and it can lead to long term relationship. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. ...

Need someone who has experience with configuring Freeswitch to detect beeps in voice mail for various carriers i.e. sprint, verizon, at&t, etc. We are effectively dropping calls in voice mail using mod_vmd on freeswitch but we are failing to detect beep so the message is playing before the beep. We need it to wait for the beep before the message starts to play in the users voice mail.

Need someone who has experience with configuring Freeswitch to detect beeps in voice mail for various carriers i.e. sprint, verizon, at&t, etc. We are effectively dropping calls in voice mail using mod_vmd on freeswitch but we are failing to detect beep so the message is playing before the beep. We need it to wait for the beep before the message starts to play in the users voice mail. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverabl...

Hello, I installed Asterisk 1.6.1.6 for a client, but I have a problem with Music On hold. If I force it through the dialplan (like, for example "exten => 999,n,Dial(SIP/21&SIP/22,,m(default))"), the caller hears the music, but when an active call is being placed on hold or transferred, nothing happens in asterisk CLI and no music is played and the caller hears a blank until it's picked up again, either by the party that put it on hold or by the person to which the call was transferred. I need help in solving this problem. Please note that my budget for this bug fix is small. Thanks, David

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PHP website + A2billing + Freeswitch installation on a fresh CentOS linux Hi @all I plan to sell VOIP minutes to local callshops under my own label [wholesale origination and termination, SIP to SIP] I ll rent a dedicated CentOS server + SSH so you ll can interact remotely. Website needed is with just basic functions : create an account, view CDR, recharge balance [i ll do credit card system myself using moneybookers or paypal API] that s all, no calling cards, no softphone etc... just a working a2billing switch + end user basic website

We are in need for someone to help in writing telephony outbound application using freeSWITCH open source platform. ? Currently we already have hosted type service and looking to migrate to new platform. I would ? like to use event socket library to talk to freeSIWTCH. This means we will run our application out of process. Our application will be in C/C++ so you must be proficient in C++ , object oriented programming. ? I do not want to use any scripting languages because this application have to perform so many other things. ? If you have worked on/using freeSWITCH then you have already smelled about the plan to use CentOS system and scale upto 400 channels with concurrent calls. We already have dynamic callflow for play and record and digit collection. The basic requiremen...

I just moved my data from old server to new linux server running dwoo, freeswitch, Centos64bit and swig. I'm getting this error: Warning: dl() []: Unable to load dynamic library '/usr/local/lib/php/extensions/no-debug-non-zts-20060613/' - /usr/local/lib/php/extensions/no-debug-non-zts-20060613/: cannot open shared object file: No such file or directory in /home/site/public_html/ on line 23 ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--D...

We have built a voice mail delivery script and need help with some refinements and beep detection. Are you available to do some configurations?

Looking for someone who can script in freeswitch. I am trying to build an application that can detect beeps for voice mail. Please send me your experience with Freeswitch and examples if you have any. ## Deliverables 1) Complete and fully-functional working program(s) in executable form as well as complete source code of all work done. 2) Deliverables must be in ready-to-run condition, as follows? (depending on the nature? of the deliverables): a)? For web sites or? other server-side deliverables intended to only ever exist in one place in the Buyer's environment--Deliverables must be installed by the Seller in ready-to-run condition in the Buyer's environment. b) For all others including desktop software or software the buyer intends to distribute: A sof...

Uncompleted outbound telephone survey app with 3 modules (Freeswitch scripting, VB6 app, SQL database) that needs to be completed. Bidder must have experience with all 3 areas. Bidder *must be reliable* and able to complete this job within 3-5 days from start to finish - not interested in coders who want cannot complete this within these timelines. Bidders must demonstrate strong ratings for communication with previous Buyers, and proven results. Preference to Bidders in Eastern Standard Time zone. Most coding for this app is done but the remaining functionality to be coded: i) recurrence function to automate recurring 1 outbound telephone survey every X-XX weeks, during user-selected start/end times for each day of the week (Sat-Sun), with option for user to change the schedul...

Looking for someone to develop, over the weekend, a quick asterisk implementation of the following functionality: Caller 1 comes into system, answered by extension B Extension B talks to user A, puts them on hold Extension B calls Extension C Extension B talks to Extension C Extension B puts Extension C on hold, goes ba...comes into system, answered by extension B Extension B talks to user A, puts them on hold Extension B calls Extension C Extension B talks to Extension C Extension B puts Extension C on hold, goes back to Caller 1 and talks again. Extension B pushes a button which puts all three into a conference Extension B can then hang up, and Caller 1 and Extension C keep talking. All we need is the dialplan/AGI code to make this happen on Asterisk 1.6.1 and Fr...

I need a script written using PHPAGI that can be called from Asterisk Dialplan. A datetime string will appear as YYYYMMDDHHMMSS. (Year,Month,Days,Hour,Minute and Seconds). 1. Function SubDate() - This function will Subtract two datetime variables and return a number of seconds, minutes, hours, days or a new datetime string. It should be able to handle leap years. Example: SubDate(datetime1, datetime2,seconds|minutes|hours|days|datetime) 2. Function AddDays() - This function will add days to a datetime string and return a new datetime string. AddDays(datetime, increment) 3. Function AddHour() - This function will add 'X' hours to a datetime string. Example: AddHour(datetime, increment) 4. Function AddMin() - This function will add 'X'minutes to ...