Kamailio freeswitch fusionpbx jobs
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.
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.
install and configure multitenant freeswitch + fusionpbx and astpp billing
- Asterisk script for incoming calls - Commenting Kamailio configuration file
We have installed test Kazoo cluster with two nodes. We needed that WebRTC properly work in Kazoo include client with IPv6 addresses so need to apply patch . Now we have some errors with jssip: SIP/2.0 488 Not Acceptable Here mod_dptools.c:3277 Originate Failed. Cause: INCOMPATIBLE_DESTINATION and sipml5: SIP/2.0 603 Failed to get local SDP - when I trying to answer on incoming call. Client has ringing but can't answer. Additional task - need to enable incoming call from external carrier. (Now I can't select option Peer in Kazoo-ui carrier settings). After fix this issues we need simple report how to solve it in feature.
Hi I have a project for scalable registration / NAT handle. This is for mobile/web/PC sip clients behind firewall to be able to register with Opensips. When the sip client...postgres DB for the credential data and it can be also cached with memchache with better performance. The sip phone will be from web or mobile and behind nat. It will use webrtc for media. The implementation should allow webrtc media to flow between each cleints. When the sip client makes a call to a real phone number as indicated by a special prefix, then opensip should direct the call to Freeswitch which will transcode to g729/g711. . The purpose of this is so that many clients can be registration at the same time with better performance and scalability. There needs to be 2 opensip running in DN...
We require to install Fail2ban in Kamailio to block certain IPs if max number of unsuccessful SIP REGISTER messages or failed login attempts (as root) is reached. If an IP is blocked an email should be sent to a target email address. We are open to another solution - PIKE if it is a better one.
Hi. I need a freeswitch installation completed with the following setup: Carrier >>>> SIP >>>>>FreeSwitch >>>>>>My SIP Server The Freeswitch will act as a either an SBC or Media Server. The carrier sends in Inband DTMF but we only accept RFC2833. The carrier will be used for both incoming and for outgoing calls and RFC2833 will have to be converted to Inband on both directions. No registration, just need to change the dtmf so if Freeswitch didn't handle the media, then it is better for me, avoid extra throughput on my servers.
Hi I need to store conference history to store the begin and end of a conference session, and begin and end of each participant. I also need to configure the following prompt: Participant Feature Keys *2 Caller Count *3 Breakout Rooms *4 Instructions - conference instructions *6 Mute/Unmute - caller controlled muting Host Feature Keys *1 Manage Q&A session *2 Caller Count - plays the number of parties in the call *3 Breakout rooms *4 Instructions - conference instructions *5 Listen only modes - host controlled muting *6 Mute/Unmute - caller controlled muting *7 Secured/Unsecured - stops callers from entering *8 Tone controls Playback Feature Keys 4 Rewind 1 minute 5 Pause/resume playback 6 Fast forward 1 minute EXPLAINED: Manage Q&A - *1 key (host only) The following comma...
Hi I need to store conference history to store the begin and end of a conference session, and begin and end of each participant. I also need to configure the following prompt: Participant Feature Keys *2 Caller Count *3 Breakout Rooms *4 Instructions - conference instructions *6 Mute/Unmute - caller controlled muting Host Feature Keys *1 Manage Q&A session *2 Caller Count - plays the number of parties in the call *3 Breakout rooms *4 Instructions - conference instructions *5 Listen only modes - host controlled muting *6 Mute/Unmute - caller controlled muting *7 Secured/Unsecured - stops callers from entering *8 Tone controls Playback Feature Keys 4 Rewind 1 minute 5 Pause/resume playback 6 Fast forward 1 minute EXPLAINED: Manage Q&A - *1 key (host only) The following comma...
we need someone to help us to develop a few functionalities using freeswitch, like a custom auto dialer system and queue api.
Looking for someone to help advise us on best Freeswitch methods and general VoIP consulting.
...Debian GNU/Linux -- The best Linux distribution! A VPS or dedicated hosting provider or a Raspberry Pi that respects your freedom! You cannot run Kamailio from behind a home network with NAT if you want to call anyone outside your home. Seriously, if you want to do this you are in for a world of advanced IP networking configuration and application code. It's possible but you really do not want to do this Kamailio -- A modular SIP router, user registration server, and NAT traversal utility...and so much more rtpproxy -- a small utility to proxy encrypted audio and video streams. Works with Kamailio to solve NAT traversal Freeswitch -- A SIP softswitch. Provides testing services like an echo test. Can also provide automated call service...
i want do an html5 gui for manage freeswitch with all features of fusionpbx but more friendly
I'm looking for a tech who already has completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid. Please include the past project info. Thanks Skills required: Asterisk PBX, Telecommunications Engineering, VoIP
Need help debugging and Migrating users to Freeswitch / Fusion PBX
I need help tuning and debugging User migration to Freeswitch Server.
I have a freeSWITCH module configured with mod_pocketsphinx and mod_managed. When I am trying to read some text using ivr_play_and_detect_speech, I am not getting proper data.
Install, Deploy, and configure a Freeswitch/PbxFusion Voip server on a OpenVZ host. 1 ip with fallover, 3 virtual extensions with softphones, custom CID.
Install, deploy, and configure Proxmox 3.4, (3 nodes) to work with voip containers, asterisk, freeswitch on an openvz host.
Need sms integrated with our freeswitch/fusionpbx cloud environment - Looking for other enhancement suggestions as well for additional fee's
Project Description Install deploy configure Proxmox 3.4.x. With contained firewall to an OpenVZ VPS push button consol for the immediate deployment of: 1. Asterisk pbx (incrediblepbx 11-12) ip 1 fallover 3 virtual extensions, 1 custom cid Install, deploy, configure 2. 1 freeswitch/pbxfusion 1p 1 fallover 3 virtual extensions 1 ip, 1 custom cid Install, deploy and configure Project ID: 8443026
Project Description Install deploy configure Proxmox 3.4.x. With contained firewall to an OpenVZ VPS for the immdiate deployment of: 1. 1 Asterisk pbx (incrediblepbx 11-12) ip 1 fallover 3 virtual extensions, 1 custom cid Install, deploy, configure 2. 1 freeswitch/pbxfusion 1p 1 fallover 3 virtual extensions 1 ip, 1 custom cid Install, deploy and configure Project ID: 8443026
Project Description Install deploy configure Proxmox 3.4.x. With contained firewall to an OpenVZ VPS push button consol for the immediate deployment of: 1. Asterisk pbx (incrediblepbx 11-12) ip 1 fallover 3 virtual extensions, 1 custom cid Install, deploy, configure 2. 1 freeswitch/pbxfusion 1p 1 fallover 3 virtual extensions 1 ip, 1 custom cid Install, deploy and configure Project ID: 8443026
Install deploy configure Proxmox 3.4.x. With contained firewall to an OpenVZ VPS for the immdiate deployment of: 1. 1 Aterisk pbx (incrediblepbx 11-12) ip 1 fallover 3 virtual extensions, 1 custom cid Install, deploy, configure 2. 1 freeswitch/pbxfusion 1p 1 fallover 3 virtual extensions 1 ip, 1 custom cid Install, deploy and configure
Installation of Freeswitch and Kamailio 1.) Install and configure 2 modules from Kazoo package (freeswitch and Kamailio + all related modules) 2.) Modules have to be configured so that they could be commutated with the separate Kazoo complex. 3.) The environment is behind the NAT mechanism, in an internal network of a customer. It is necessary to realize opportunity to use a local traffic in an internal network of a customer ,and to use a global traffic only for calls to subscribers out of a local network. 4.) Also on an environment has to be configured an firewall so that access to the server will provide only on necessary for work of a telephony, and service ports. Environment: Distributor ID : CentOS Description: CentOS release 6.7 (Final) Rele...
We need a simple power/auto-dialer system that follows this flow: - Manager will upload via Manager interface a list with these fields: Contact lead list: Name, company, title, phone number, email - Manager can setup via Manager interfaceX number of agents where each agent can call from a uploaded list. - Agent can access the agent interf... We currently run 3cx software on VOIP at our offices and above add on program to be comapatable. 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. If you have a pre-made solution or know someone please advise. Would prefer a Freeswitch solution, but Asterisk would b...
...would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about it and have done it before. We have got many bids people saying "I can do it" and this is not enough. We definitely don't want solution in your response but your description has to be detailed to gauge the potential of project. We will provide full assistance with programming and development in case you are not expert enough. Here are implementation details : - Operating system CentOS - Telephony platform FreeSwitch - Must need to provide script details (Any language will work) if external interaction is needed. - Must provide C-language module if tight integration is needed with FreeSwitch - Solu...
I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read ab...voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch
I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read ab...voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch
I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read ab...voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch
I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read ab...voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch
I need to integrate Fusion PBX with ASTPP Only bid if YOU HAVE DONE this before Please bid, and be ready to show me that you have already integrated. If there is no demo of an integrated system as required above, then DO NOT bid. Only bid people who have experience with ASTPP and FusionPBX ** We need to implement an ASTPP billing box being a trunk for fusionpbx, later we will connect more fusionPBX servers ****** PLease No Milestones would be create until the project is completed ****** ****** Money Would Be Released When The Project Done and Full Test ******
I need a Lua script example that 1. Plays music on Leg A 2. creates a new session - orginates the call 3. on Answer wait 3 seconds 4. DTMF on Leg B 5. Bridge call 6. Stop music playing on Leg A
I need a Lua script example that 1. Plays music on Leg A 2. creates a new session - orginates the call 3. on Answer wait 3 seconds 4. DTMF on Leg B 5. Bridge call 6. Stop music playing on Leg A
Need to Setup and Configure Asterisk
Set up test accounts, registrations, gateways and LCR on Siremis/Kamailio. Server is up and running, network in place, but we have no experience with Siremis and would like to have it set up so that we can build production form test bed. Looking for experience with Siremis
I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read ab...voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. Please bid only if you know about and able to help. Thanks Desire Skills Freeswitch
I need an api setup for free switch. I will be load balancing freeswitch boxes using Kamalio and I want call control through API's. I need to be able to do a lookup by incoming phone number to determine which freeswitch box is handling the call, and then send call control commands via an API.
...will serve as basis for end-to-end communication solution. With the right skill set and attitude you will join a young and motivated international team, working on the design and implementation of the opensource based server infrastructure. Your main responsibilities will include: * Design and development of a VOIP/SIP module within existing opensource SIP Server such as OpenSER, Asterisk, Freeswitch, etc... * Debug and analyse functional and performance issues within existing VOIP communication platform. * You will be challenged with identifying performance and scalability bottlenecks * Analyzing communication protocols like SIP, RTP, JSON. * You will also work with the rest of the team to ensure compatibility between the different server components, mobile and web cli...
Need some Functions Keys created for Intercom /Paging
Freeswitch has been currently runs in AWS EC2. but there are some configuration need to be done 1. Configure Freeswitch auto-NAT 2. Configure linphone SIP App android and linux app with two SIP accounts 3. Register two SIP account to freeswitch via 4G and WiFi 4. ensure two apps can call each other via 4G or public WiFi No software coding is needed. If you have done it, you know what it is.
