G723 freeswitch työt

Hi, I need to have someone setup Freeswitch / Plivo () and configure it to work with a live SIP trunk that I can provide. I will also provide root access to a CentOS server. Final outcome should be that I can make and control an outbound call from a simple PHP script (similar to how Twilio works), and the call will be made using the Freeswitch/Plivo server. To do : - Install Freeswitch - Install Frameworks - Configure outgoing carrier for outbound calls - Configure incoming calls redirected to a PHP file located on another server Need this done ASAP (within 48 hours) Thanks!

Hi fellow Bangladeshi!nnI need someone to develop a freeswitch lua dialplan that I can drop into an existing vBilling system replacing their and files. These lua files are compiled and protected against reverse engineering/decompiling (it seems). So I need lua dialplans created from scratch, that use xmlcurl. Functionally, it should achieve as much as the current bin files do except the call limit settings...nnI am able to:n1) supply full access to the server for you to test and develop onn2) give you tcpflow output when a call arrives to the switch in various scenarios (eg rejected call, accepted call) so you can see the sql run for you to replicate...nnThis is an urgent job and needs to be worked on fulltime for completion in 2 weeks.

IPLINK is looking for high level and experienced Asterisk and/or Freeswitch Developers for a number of projects. Projects include high demand internet radio streaming, conferencing, and chat (accessed by the PSTN using DID's). Developer must have extensive experience working with these applications and proven track record. Additional sills required: Extensive database knowledge (e.g. mysql), good UI development (for internal management, not customer facing), project management, telecommunications, and VOIP. Developer must be proficient at speaking English.

I need a program in Freeswitch/windows that can dial from a list of numbers and report back to a database the results.

Objective is to configure an existing test-installation (no traffic) of Freeswitch on an Amazon EC2 instance. You need to be proficient in the Freeswitch Dialplan XML and also in Python as the server is running a Python script for DNIS manipulation based on inbound IP address and other criteria. You also need to be familiar with Arkadi's work to build an IPP based codec lib for this server to allow transcoding. Please don't bother to bid on this if you haven't done this successfully at least once before or at least know EXACTLY what this is about and how to deal with it. As "proof" please mention the sentence "Yes, I have" at the very beginning of your bid. If you fail to mention that your bid will be ignored. You will need to provide...

I need Auto Dialer based on Free Switch. Just installation and Configuration. Please send me your quote and expertise.

Configure Kamailio to route calls to destinations in Asterisk and Freeswitch and to PSTN. Kamailio 4.0 is already installed, but the configuration needs to be fixed. Establish 3 sets of gateways - Asterisk - Freeswitch - PSTN Route to Asterisk and Freeswitch gateways based on DnID (round robin within each gateway set). Route outgoing calls to PSTN group of gateways (lcr). Always check that the call is from defined gateways

...termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Other standards codecs like G722, G726, alow, ulow... shall be supported - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels if they are busy. - If a channel is busy the call shall try automatically the next channel and so on till it finds a free channel. - FreePBX shall...

...etc. we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may server...

Asterisk has support for Answering machine detection however it does not have support for Voicemail beep detection (as I know) like freeswitch module named mod_avmd therefore we need script / code to detect voicemail beep in asterisk Kind Regards

...etc. we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may server...

Asterisk has support for Answering machine detection however it does not have support for Voicemail beep detection (as I know) like freeswitch therefore we need script / code to detect voicemail beep in asterisk Kind Regards

We need to configure FREESWITCH to be able to handle phone calls. We are using BiglueButton which is already integrated with FREESWITCH. So ultimately phone calls will be handled in Bigbluebutton through FREESWITCH.

I'm looking for Softphone Dialer for windows and Mac work with all codecs ( G723 , G729 , G711alaw , G711ulaw ) something like

I had a freeswitch module developed about 2 years ago written in C and supporting python scripts. Now I need to add more functionality to this and maybe get rid of the python scripts and do it in C.

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web based Calling card and call forwarding solution required using asterisk /a2b or some other open source switch like freeswitch . Preference will be given to somebody which has already done similar project . We can send reference calling card website . need to get it done asap.

... we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may s...

Utility to send a fax (from TIFF image) using the T.38 protocol (or G.711 passthrough), over already established SIP call session. Very similar to 'spandsp_util' described here It must work preferably on Linux server (Windows server is also an option) and must NOT use any switch software like asterisk, FreeSwitch etc. or hylafax software. Mast be capable of transmitting 250 simultaneously. Command line parameters will be passed to it: -f TIFF file name -r remoteIP:remoteport -l localport -i unique id -p T38 or G711 -t TSID ATTENSION: Milestone will be released only after successful transmission of fax.

The project consist of building a .Net based windows Web GUI for administration of the Freeswitch service. to build a windows .net based VOIP PBX. the freeswitch open source project can be found at: Features Extensions Auto Attendants Hunt Groups Conference Rooms ACD Queues / Call Center Paging Call Recording PnP Phone provisioning Fax2Email Barge, Listen, Whisper Trunk Accounting CSTA WebRTC Domains/multi-tenant Service Flags IVR Nodes

The project consist of building a .Net based windows Web GUI for administration of the Freeswitch service. to build a windows .net based VOIP PBX. the freeswitch open source project can be found at: Features Extensions Auto Attendants Hunt Groups Conference Rooms ACD Queues / Call Center Paging Call Recording PnP Phone provisioning Fax2Email Barge, Listen, Whisper Trunk Accounting CSTA WebRTC Domains/multi-tenant Service Flags IVR Nodes

Hello We are looking for an expert on He must be expert in FreeSwitch as well. Who can install it on our servers, customize it according to our needs, give custom look according to our company design and look. Thanks and Regards

I need to integrate an IP PBX (FreePBX) with BiBlueButton application that uses FreeSWITCH as a voice platform. The idea is that all branches of the company can connect directly to the conference Web without a computer. More information visit: # Does_BigBlueButton_use_Asterisk_or_FreeSWITCH_?

I need an answering machine detection for FreeSwitch to integrate our outbound calling app in Linux.

We have a need ...news also have running ticker. Data Encryption (VPN) When the dialer communicating with server , packet should be encrypted and packet will be decrypted in the server. The decryption of sever side also has to be done from the same developer. Also the same encryption must be configurable for router (DD-WRT, Tomato or a custom ISO) for VOIP adaptor use. Other Features • Support G729 & G723 codec • Credit transfer among users • Facebook integration • Integrate facebook chat with messaging module • Manual sync of address book • Select options wifi only/wifi,3g • Should optimize the application for maximum battery use Minimum of three months warranty needed for all the apps , all bugs must be fixed within ...

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We have a one to many VoIP broadcasting system that can be accessed through a flash based client. We needs to make further improvements in it's functionality. We need urgent help with the development. Please give us a time/cost estimate based on our requirement. We would like to extend the project if the current tasks are handled well.

We need a softphone You will develop softphone and make it available in the respective stores (app store, android market etc) We wo...following platforms. IPhone - must Android - must Blackberry - optional but prefered Windows Phone - optional Nokia - optional Samsung - optional When you bid you must list what platforms you can develop for. Phone will have our logo and our sip server will be hard coded. user will enter their login and password and the softphone will save this info. I am very interested in having g723 and g729 if you can advise on this. There will also be a manage page for other functions, we would like this manage page to be accessible by clicking a link from sofphone. The mange page uses the same login as softphone so we would like the logins t...

...Cool Asterisk/FreeSwitch/ViciDial Programming Project (Amazing Opportunity) IMPORTANT NOTE: We hire freelancers many times a day. We are ramping up for MANY new projects and looking to get some fresh people into our talent pool. There is a lot more work for you if you do a great job. A LOT MORE. We are very comfortable working with individuals in India, Pakistan, Eastern Europe, Asia/Pacific, Central and South America, and so forth. We have project managers on staff who speak Russian and Spanish. Project Details ***Due Date: 30days, but with individual milestone components being completed earlier and at regular intervals. ***Job Description This project is for a lead management system for call center dialing data. It should integrate with Asterisk and FreeSwitch...

We have web dialer application based on plivo+ freeswitch. Freeswitch is going down while oubound dialer campains are running (about 120 outbound calls per minute). We do have auto restart for out cloud server (plivo+ freeswitch) that someone which is no longer avialable did but still want to know what happend and to fix it. we need freeswitch/plivo expert and web developer in order to fix our fusionpbx auto scripts. If everthing will work fine we have a lot of jobs to do and we are looking for someone trustable for long term

...provide you the Dedicated server asterisk and client asterisk as for model. configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may server...

freeSWITCH for GSM Termination with (GoIP) skype:

freeSWITCH for GSM Termination with GoIP

I have Goautodial setup and need help configuring to allow the following processes: - Agent escalates a call to a Manager. Manager is able to see Customer information that Agent was viewing. - Agent escalates a call to Manager by conferencing in the Manager. Customer should not hear any ringback tones, or any indicator that Manager was brought into call. - Manager has ability t...hear any ringback tones, or any indicator that Manager was brought into call. - Manager has ability to specify whether he is available or not to allow Agents to add him into a Customer call. - Manager is able to make outbound calls to Customers and at the same time is able to receive escalation calls. Thanks for your help. Keywords (ignore): Goautodial, Vicidial, Osdial, asterisk, free...

We are a little 4 man company providing IT-services for our customers. Lately we developed plans to provide VOIP services too, virtual PBX to be sp...Call-center Solution,...). 4. is interested in a long-term contract and close relationship. We are not looking for a onetime project. The on who designed the system, should maintain it. All technologies attached to that project are required (SMS, Billing, Faxservers, Protocols,...) We want to start in the next 2 months. We are specifically looking for multitenant specialists (Asterisk or freeswitch). We DO NOT look for someone who can setup a single Asterisk server. If you have have any questions about this job please feel free to ask us. Give us an idea how you would achieve that in your first contact and we will p...

Hi We need a perl regex to check a logfile for an entry [WARNING] sofia_reg.c:1402 SIP auth failure (REGISTER) on sofia profile 'sipinterface_1' for [] from ip "tralalalala" The sip interface should be catch all, so if sipinterface-1 is sipinterface_2 it should also work, Its for csf rules for an asterisk server are there but we use freeswitch # Example: # if (($lgfile eq $config{CUSTOM1_LOG}) and ($line =~ /^S+s+d+s+S+ S+ pure-ftpd: (?@(d+.d+.d+.d+)) [WARNING] Authentication failed for user/))$ # return ("Failed myftpmatch login from",$1,"myftpmatch","5","20,21","1"); # } # # The return values from this example are as follows: # # "Fail...

Hi, I am looking for a developer who can provide us VoIP Dialer for Android, iPhone, (From Open Source is no problem) The Dialer should not have any developer logo or any other link. The IP address should be changeable i want it to work on Codec : G711 For Free , to use G729 and G723 its a payable future to use 3G, 4G and VPN also is a payable future On Main Screen user should able to see: ~ Keypad - Register or unregister ~ Last Dialled Numbers ~ Access to Phone Contacts. This should support all version of Android And iPhone Thanks

...demonstration # We will provide you the Dedicated server asterisk and client asterisk Configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) Continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; Continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may ...

Looking for someone to create situation in which csf will check all appropriate logfiles and take appropriate actions to protect our system. We have a freeswitch server which we want to protect from wrong logins. Its simular to fail2ban We can not provide a test enviroment so u would have to have your own freeswitch server or have enough skills we will test the config on our server and when its working and thus catching failed logins on all services, we reserve the right to ask for one or two extra logfiles to check

I need Newfies and Freeswitch installed and configured on my server. I will be doing voice broadcast and sms broadcast campaigns. I have voip providers already. Thanks!

In your bid, please describe your proposal. Thanks. We need a simple auto-dialer system that follows this flow: - Admin will upload a list with these fields: Name, company, title, phone number - Admin can setup X number of agents where each agent can call from a distinct uploaded list. - Agent can access the agent interface by logg...will be able to see the Name and Company of that contact. The above are all the features we need (pretty simple system). If you have a pre-made solution that you'd like to sell a license to, or know of any open source platform that can accomplish this, please bid. This is a simple application, and we need it done urgently. If you have a pre-made solution or know someone. We prefer a Freeswitch solution, but Asterisk would b...

I have a bluebox vps and an elastix vps. Elastix has trunks to reach to the outside world as well as receiving incoming calls from the outside world. Elastix is also configured with a2billing billing system to auto calculate minute usage and other billing related things. Because a2billing doesn’t work on freeswitch, I had to use Elastix, but Elastix doesn’t support multi-tenant, so I created another vps and I installed bluebox on it. I would like bluebox to use Elastix as a trunk to reach the outside world. In other words: Bluebox ==> Elastix ==? Outside world…….. as well as……… Outside world ==> Elastix ==> Bluebox Again the main reason for my above requirement, is because I need a billing system for bluebox, and I ...

...etc. we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can measure the BW compression making the SIP-> IAX call trunking, need develop a simple WEB tool to change IAX IP and port (you understand that it is sensitive option when trunk is blocked by country border GW) continue building up main server with codec conversion (will install g729/g723 codecs) amd Install OpenVPN Server&client - at this stage we will test it and measure the BW compression with all kinds of options like codecs and openvpn compression modes; continue project with compiling the automated installation distribution (with OpenVPN, Asterisk, Codec conversion, IAX trunks config ) for client-side CentOS system, which can be distributed to may ...

IGNORE budget and bid what you believe is a fair price I have a server with the following specs: 2U server 2 quad cores 16gb ram 10tb hard drive space It is an empty server with nothing on it. I will have 5 agents making outbound calls. I need someone to help me setup any platform that would allow for the following: - An admin to login and upload a CSV file (co...empty server. I'd like to use ONLY pre-existing already-developed platforms for this. Also we will need to have a CRM integrated with the platform so that leads called can be sent to this CRM. We are open to any opensource CRM (sugarcrm, vtiger, etc.) Need this done by the 5/26 Keywords (please ignore): CRM, power dialer, powerdialer, auto dialer, autodialer, predictive dialer, freeswitc...

Need an Asterisk / FreeSwitch expert to create a phone system similar to to spoof caller ID's. Please see

Need an asterisk/freeswitch expert to create a virtual phone system similar to or onebox.com. Some but not all of the system shall include: -call forwarding / call routing to different numbers similar to google voice -dial by # extensions -auto attendant / "music on hold" -greetings -ability to conference call -voicemail with voice to text -and a few more features to be discussed at a future time. PLEASE CHECK AND TO ENSURE YOU UNDERSTAND THE SYSTEM AND FEATURES WE REQUIRE. DO NOT BID UNLESS YOU HAVE UNDERSTAND THE SAMPLE WEBSITE AND KNOW HOW TO CREATE A SIMILAR SYSTEM.

I need of a programer that knows freeswitch and knows how to write lua scripts.I have a script already that does conversion for pdf to tiff ,I need a script for text to tiff and doc to tiff. If php knowledge that would work also. This project needs to be completed today no exceptions..

Good Day, I am looking for a MultiTenant solution based on Freeswitch. I will require the following - Web based Interface - Hosted PBX Module - SIP Trunking Module - Conference Calling Module - Session Border Controller - Kamillio or OpenSIPS - Provisioning - SLA Support - DNS Setup - FreeRadius Setup for PPPoE authentication - Receptionist Console (Web Based) - End User Application (Web Based) - Server Health - Deploy additional Servers - Automated Backup - Ability to install a freeswitch unit on the customer premise to allow local registrations and survivability, but have it controlled and managed from our main PBX interface. I need to able to create a Tenant and select specific features I want to assign to the tenant, such as Hosted PBX or SIP Trunking...

Poszukuję osoby posiadającą doświadczenie w konfiguracji platform FreeSWITCH i Asterisk oraz telefonów Polycom. Rozliczenie godzinowe.

We have web dialer application based on plivo+ freeswitch. Freeswitch is going down while oubount dialer campains are running (about 120 outbound calls per minute). We do have auto restart for out cloud server (plivo+ freeswitch) but still want to know what happend and to fix it.