Nelly codec sip rtppekerjaan
we are looking for an expert in persuasion and relationship with voip company to get you the way route c li a to z Required information : *route cli a to z 100% *sip *Allow miss call The proposed companies: *tel... *sky *pccw *idt Subscription type Annual or monthly without Note:This work can be permanent or incentive if all requirements are met
I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: You will probably also need to check the G729 codec configuration
Hello, we have a freshly installed Goautodial server, but we have some issues with the dialplan entry. We want to have outgoing calls with a fixed +31 number, all the details (SIP/DID) are already available,
we are looking for an expert in persuasion and relationship with voip company to get you the way route c li a to z Required information : route cli a to z 100% sip Allow miss call The proposed companies: tel... sky pccw idt express Subscription type: Annual or monthly without Note:This work can be permanent or incentive if all requirements are met
Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (Dinstar gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from US...
We are trying to build a new Asterisk server using PJSIP instead of the older SIP module and need some help to make it work. The setup is as follows: Asterisk certified/16.8-cert3 running on CentOS 8.2 with a public IP address. Phones in the office need to connect to the Asterisk server, they are on a private LAN (192.168.1.x/24) and access the Internet via a NAT router. Two types of phones: Aastra 6757i and Grandstream N300. The server also needs to send and accept calls over a SIP trunk to another Asterisk server also on a public IP (PSTN Provider). You will need to be an expert with Asterisk and PJSIP and work with us to get this working and explain what you did - you can have SSH access to the Asterisk server if required. For the right person this should be a fairly...
- Must be compatible with PBXAct and FreePBX. - Can run in background and stay registered even if multip...registered even if multiple programs are open after. - Be able to Forward a call to an internal extension or to a phone number. - Be able to do a Conference call. - Have a Contacts list. - Compatible with UDP/TCP/TLS for Transmission Protocol - Can register multiple account at the same time - Support multi-language (French and English) - Support RFC2833 as DTMF - Support PCMU (ulaw) as Voice Codec - Call History And a (nice to have) : - Use a QR code to add an account easily - Multiple ringtones - Support QoS - Import contact list from iPhone, Androïd or Outlook. - DND option If you can give me a time frame to build the basic and another tim...
...BUN-K9, with AIP-SSM-20. SIP trunk with 2 DID, 4 lines. Features: Redial (Included) PickUp (Included) GPickUp (Included) Caller ID (Included) Intercom (Included) Call Transfer (Included) Group Paging (Included) Music On Hold (Included) Do Not Disturb (Included) My Phone Apps (Included) Call Forwarding (Included) Company Directory (Included) Personal Speed Dials (Included) Hunt Pilot, Call Hunting (Included) Over 37 Custom Ringtones (Included) Internal Call/External Call Ringtones (Included) NTP Synchronized Date, Time, Events (Included) Individual Named, Numbered Extensions (Included) Directory of placed, received, missed calls. (Included) Ad-Hoc Conference from VoIP Phones to Any Phone (Included) Inbound/Outbound Calls From All Phones and Extensions (Included) SIP Trun...
Create a simple VOIP Application using React Native 1, Must use latest React native version 2, Should able to work with existing SIP Server (FreePBX and VitalPBX) 3, Should able to make an outgoing call 4, Event Trigger when DTMF is received.
simulation network for video streaming using codec H.265 using Matlab
expert in omnet++ to extract a dataset for video streaming with codec H.264
I need a panel for my business and call routing sip trunk
...kategori besar yaitu: a. UCaaS (Unified Communications as a Service) b. SIPTraaS (SIP Trunk as a Service) c. CCaaS (Contact Center as a Service) d. Network and Security as a Service (by CATO Networks) a. Layanan UCaaS atau Layanan Teleponi (Voice, Video, Messaging) - Cloud PBX Untuk pengguna yang tidak memerlukan perangkat PBX di perusahaan. Akses layanan dapat digunakan melalui ponsel mobile apps di ponsel berbasis Android, IoS (menyusul), melalui PC/laptop Windows atau Mac, dan melalui IP phone, sehingga user experience yang dirasakan adalah Fixed-Mobile Convergence Mendukung fitur umum PBX seperti IVR (interactive voice response), call forward, parallel ringing, dsb b. Layanan SIPTraaS (SIP Trunk as a Service atau Cloud Trunking) Untuk pengguna yang mem...
We are Looking for a freelancer who can developer and AV1 Encoder from GITHUB and Decoder as well. Purpose is to take a real Live Feed from NVidia or any Other Card or IP input as Uncompressed Video to Compress in AV1 Codec Encoding Method should be untestable by the Developer as we will required some modification . Decoding path the reverse Engineering of Encoding We can use Nvidia GTX 30 Series Card. Looking forward to get some Experience Developer.
We need a person who is knowledgeable with Pfsense firewall, Freepbx and AT&T trunking. I have installed freepbx server. It is behind the pfsense firewall, and located on the LAN part of the firewall. We are having difficult time to connect to ATT VOIP sip servers with 2 trunk IPs and 1 SIP signaling IP. Our time is limited so we need urgent help. Can you help me with this issue?
We need a person who is knowledgeable with Pfsense firewall, Freepbx and AT&T trunking. I have installed freepbx server. It is behind the pfsense firewall, and located on the LAN part of the firewall. We are having difficult time to connect to ATT VOIP sip servers with 2 trunk IPs and 1 SIP signaling IP. Our time is limited so we need urgent help. Can you help me with this issue?
Omnet ++ to extract a dataset for video streaming with codec H.264
Need to switch PRI lines to SIP need any expert to help me , will have manager on remote if need be 2 locations , need to be switched from PRI to SIP protocol
To extract data set from omnet for video codec H.264
We want to have sip trunking between Avaya and asterisk pbx.. If anyone have experience and stays in Jeddah let us knw
I am looking for a Windows 10 Listener Service / Azzure Service that communicates with Teltonika devices using codec 8. The solution must also receive and transmit setup TCP messages from/towards the trackers. Please contact me for more detials
I would like to buy a tcp listener to communicate with teltonika Codec 8. Please contact me to discuss requirements.
...We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to WhatsApp gateway 2) whatsapp gatew...
I need VOIP audio chat Client/Server. Need to manage channel and users, and must work on linux and windows, use OPUS codec, and low latency. Configuration must be into external config file.
We have a requirement to assist with the initial configuration of an Audiocodes Mediant 2600 SBC to connect a Gamma SIP trunk to an Avaya IP Office PBX and guidance on setup of dial plan routing. Must have Audiocodes skills.
...redundancy. We have attempted without success to upgrade our current setup to OpenSIPS 3.1 as a cluster for redundancy and load balancing. Our current setup is single public IP address using a destination NAT rule on a Mikrotik Router to route SIP traffic to a single OpenSIPs instance. All servers are running on Proxmox LXC containers. OpenSIPs is using a MariaDB cluster along with RedisDB for storing data. We need to be able to fail over from one server to another for maintenance or emergency while keeping user location (SIP Registration) and SIP dialog information in sync between both servers. When we change the NAT to point the public IP from one node to the other we need inbound and outbound calls work reliably. The current configuration is keeping this inf...
Custom Themes for Vicidial and Call flow changes for asterisk for Inbound Call Centre SIP based Outbound Call Centre Sip based IVR UI based Predictive Dialer Pls share the demo of your work on Vici Dial
Create the Softphone with WEBRTC. SIP Connecting with API Back end with get information credentials sip. Chat communications. Video Call using jannus. Screen sharing functionality during the meeting Design application the attachment.
I have a customized linphone ios and : 1- linphone log option to identify error but Its not working when sip stack crashes it should send back logs. 2- I need to change iphone registration port to 7071 now 5060.
APLICACIÓN SIP CALLING QUE REEMPLAZA LOS TELÉFONOS DE LA HABITACIÓN DEL HOTEL Y AÑADE NUEVAS FUNCIONES. El objetivo del proyecto es crear una aplicación de Android/iOS para llamadas SIP. Se requiere reemplazar / eliminar la necesidad de teléfonos de habitación de hotel con esta aplicación de llamadas. La aplicación necesita poder identificar la latitud y la longitud actual de las personas que se registren (para hacer exclusiones de zonas que no están permitidas) y tener una función para que el cliente del hotel ingrese un PIN específico proporcionado en la recepción del hotel (este PIN se genera en el hotel y dispone de una fecha de activación y una fecha de fin). Una vez conec...
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We are looking for a mediator and public relations expert to communicate with telecom companies to get SIP to raise international calls via VoIP protocol with the CLI feature
We are looking for someone capable to brand and compile SIP dialer (Android, IOS, PC) like this one: I think the best choice would be basing on Linphone code. The most important feature is ofcourse calling, phonebook should be divided to onnet and offnet contacts (onnet user list available over API). Onnet users should also be able to chat with one another, signup (new users - api available) and check tariff rates (API), Service is hosted on VoipSwitch platform. We can provide all needed technical information or create additional - dedicated API for easy development. We prefer someone who already has such released in porfolio.
I am looking for an audio , SIP , WEBRTC, EXPERT to build a product like with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all the bullshit. I will provide sample layout of homepage and dashboard to NOT CONTACT ME IF YOU DONT HAVE SAMPLE OF SIMILAR PROJECT TO SHOW ME.
we need a small app that re rout the audio of an outgoing calls in android to a embedded sip client
I am looking to build a product like with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all the bullshit. I will provide sample layout of homepage and dashboard to build
...instant message, schedule or add to a List. For each click on a number, you must save the url of the original browser. Create list for whatsapp You can create a list The list will have data accessed by clicks or lists The list can have a PLAY, which can send messages sequentially to the numbers in the list. The telephone system must be integrated with my company's telephone system. Information like SIP should not be visible. When you click on a number to call, you must call immediately. But you can also schedule the call. But you can also join a list. At the end of each call the user can make a tab. The system will have standard guides, but the customer can register guides. Connecting the playlist It will execute the calls one by one and each call answered The data captu...
We are run a software company and i want to develop a Interactive Voice Response System (IVRS) so i want to person for specialty for Interactive Voice Response System (IVRS) & sip trunk vs pri. so any personals in this line please contact me
...experienced software developer and an engineering manager, but I have no experience in Asterisk or PBX. This project is to run Asterisk on a new Ubuntu/20 server running on AWS, listen on SIP calls and handle a few basic operations: 1. Voice greeting, coming from mp3 or other audio file. 2. Connectivity to Google Voice Recognition, listening for next commands 3. Logging the recognized voice in text and in audio (mp3 or other audio file) 4. Responding with another greeting from mp3 files, depending on recognized voice (3 options: option A option B or unrecognized) Also, we own 2 VOIP DIDs in , which has SIP connectivity and DID POP. We would like to connect those, so that calls to these VOIP will reach the Asterisk service. Other requirements: 1. The outcome for the proje...
...currently are using asterisk version 11.22.0 We have BLF currently working Asterisk Feature Busy Lamp Field (BLF) so it is working but we have to create an BLF subscriber extension like this [BLF] exten => 1000,hint,SIP/${EXTEN} exten => 5000,hint,SIP/${EXTEN} exten => 5001,hint,SIP/${EXTEN} exten => 998899,hint,SIP/${EXTEN} exten => Prox1,hint,SIP/${EXTEN} But this is working. when the command asterisk -rx 'core show hints' run it shows hints for each extension This works but we want to create a simpler BLF BLF subscriber extension like this [BLF] exten => _XXXXX,hint,SIP/${EXTEN} But this is not working. when the command asterisk -rx 'core show hints' run it does not show hints for each extension ...
Hello Guys, We are looking for a consultant for SIP Truck setup in India. Consultant must very good understanding with SIP Trunk and other related technologies such server configuration/load balancer setup/API.
I need to use an android cellphone as an asterisk channel/Gateway. This will mak...working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP ...
...2. we already build a unique WebRTC C++ client (Windows 7/10) that captures the host desktop & windows and streams in real-time to standard browser peers and even act as remote controller for those remote browser based participants. 3. seeking for highly capable C++ expert to extend existing functionality with advanced cool features such as: a. supporting latest WebRTC 87 library (with latest codec supports, improved video adaption features etc') b. ability to stream host mic/system audio/camera and capture and present remote A/V streams (alternative to our existing js based implementation) c. record single host entire screen + entire A/V streams locally to mp4 file or server mode (record entire room with all participants A/V/SS streams) while managing previous recor...
Having financial consulting website wherein require loan as well investment calculators, like Personal Loan Calculator, Home Loan Calculator, SIP Calculator, PF calculator etc. Can share example which can allow to work same in line.
Need someone to set up an asterix server for us; this would include connecting to the outbound SIP provider (twillo) setting up the GUI, some pilot groups, conferencing groups, automatic call recording, call transfer, sms response to incoming calls that are unanswered, a Grandstream DP750 endpoint, some soft phones, and probably a few other things on a Debian install this would also include some asterix education along the way .. I can provide SSH access for the system at a clean install point
we need to 1. add our logo and pre configured sim domain 2. insert an api that will update balance the open source is at the budget is 100
Program a SIP trunk on a ipecs 50a phone system
