Sip trunk avaya trunkPekerjaan

I want to set-up internal communications through a PABX or a Dialer that supports SIP lines. Need the following features: Leads Management Call transfer Mobile Calling facility Live dashboard Call recordings Reports and analytics Real-time notifications Missed call services Dynamic call flows Please let me know if someone can help me out with this. The person needs to be located in Mumbai. Thanks.

App Like Digital Showroom. (Relipay, , Fintech, AEPS, DMT, Travel, Tourism, Recharge, BBPS, Pan card, Bank account opening, G2C Services, Insurance, Loans, Credit card, SIP) Location: Delhi, Noida, and Ghaziabad

I need a Panasonic NS700 telephone system factory resetting and then reprogramming with 6 SIP trunks for 6 total users. The system is located in Nottingham.

We would like to set up a SIP trunk and connect to Fusion PBX behind a PF Sense Firewall. We would like to configure incoming and outgoing calling.

Need a small design work done. Have a reference image a similar design need to be created with specifications. Key Pointers:- 1. There needs to be 5 kidneys on each side. 2...design need to be created with specifications. Key Pointers:- 1. There needs to be 5 kidneys on each side. 2. 1 kidney on Top 3. You can use 3-5 trunks rising from base rectangle. 4. Kidneys can be of size - 6 inch to 14 inch 5. Total artwork need to be of a dimension of 5ft-6inch height and 4ft width. 6. Colors - Orange and its color palette. So you can use pastel Blue, Green as well for trunk. Orange + one color 7. Kidneys don't need to have a border. 8. Inside kidneys we will have text and small icon, which will be shared once we have base design ready. 9. Reference images are attac...

...save power. Today, Flexisip supports native push systems of Android, iOS and Windows Phone ; but can also delegate the work of sending the push request to a tier service using HTTP GET Flexisip software now comprises three modules: proxy, presence and conference (the latter being required for Linphone's group chat features). This server suite is typically suitable to deploy your own cloud SIP service tuned for your Linphone-based application, especially since it supports push notifications (including iOS13 new requirements) Push Notifications PushNotification module allows Flexisip to wake a liblinphone-basd application up when a chat message or call invite cannot be delivered because the application is unavailable. This feature has become crucial since mobile OSs got used...

I need help finding a free or low cost app that I can run a virtual number through so my virtual assistant can answer calls. Any help would be appreciated and I can pay

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.. you would be combining and into 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. I will provide sample layout of homepage and dashboard to

...solution must be server friendly, we prefers it to be linux/web oriented but won't mind if its windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better). 2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple callers id which to be round-robinly used for outgoing calls, if caller-id is defined for the sip-account then will only send calls using that caller-id instead. 3. We would be able to define "What to do, when a called is Picked" means, if a call is connected, we would be able to define if the system is t...

...Cisco CUBE. My SIP provider is sending the INVITE package with wrong sip number as you can see bellow: Received: INVITE sip:765617@ SIP/2.0 (should be 0893540111) v:SIP/2.0/UDP ;rport;branch=z9hG4bK7cvF8Zyv4U9yD Route:<sip:765617@> Max-Forwards:65 f:<sip:0477770613@>;tag=4rDev7jZe34jH t:<sip:0893540111@sip> i:9f452181-e17c-1239-eabe-00163ea67143 CSeq:31664763 INVITE I've been trying to apply Conditional SIP Profile to change this header with the TO number listed bellow. (DOC I've been using: ) request INVITE sip-header SIP-Req-URI modify ".*@(.*)" "INVITE sip:u01@1" request INVITE sip-header To copy "sip:(.*)@" u01

Mitel AX Controller SX-200 needs programmed to work in a apartment complex scenario. About 140 ports/rooms lines that will get a direct phone number (number rings stating to the room) and 5 SIP phones for back office/management, no additional feature needed for units phones (like messaging or so on). Connection is via PRI and the provider is COX

...solution must be server friendly, we prefers it to be linux/web oriented but won't mind if its windows based application as long it can work with 1000 sum ports with 128GB windows server (if more then better). 2. Multiple "Caller ID" to be defined when adding VOIP/SIP/IAX lines, if no caller-id is is set, then when a call is sent thru this line, the system should get a random caller id from a list and use it, in the list we will provide multiple callers id which to be round-robinly used for outgoing calls, if caller-id is defined for the sip-account then will only send calls using that caller-id instead. 3. We would be able to define "What to do, when a called is Picked" means, if a call is connected, we would be able to define if the system is t...

Hi, I have finance blog in Hindi and want to have following calculators for my website: a. Step Up SIP Calculators b. Target Amount Calculator c. Goal Based SIP Calculators d. Provident Fund Calculator e. Retirement Planning Calculator f. Asset Allocation Calculator Thanks, Rahul

We have a web based IP phone that utilises version 0.14... we need the javascript code changed to utilise the latest support version of , which is 0.16, there is a fundamental change in the API interface and without spending more time than available, we cannot make the changes and get them to work. We would provide copies of the current source files and a test SIP username/password so you can fully test and demonstrate to us it works with the new code before we update our current production code. If you do not have experience of version 0.16 and above, please DO NOT bid on this project. Thank you.

I have 2 switches connected togather using vlan trunk and another connection going out to a firewall, traffic is not flowing out (can't ping firewall) need help from someone who understands vlans. switch 1 (vlan 1: (mgt) vlan 2: external) - port 48 trunk port switch 2 (vlan 1: (mgt) vlan 2: external) - port 12: connected to firewall LAN port 48: trunk port

We are a 4 year old telecom company and we’re looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS React These skills are optional: Facebook APIs We will offer long term work every week but only if you prove your skills in the first week or so. We are only looking for serious developers and to prove that please fill the attached document, upload it anywhere and add the link to the project proposal. Your project proposal sh...

Ich habe einen MacOS Catalina und benötige Hylafax um Serienfaxe zu versenden. Auf dem Mac soll das Paket T38Modem kompiliert werden unter root per SSH Zugang für Sie remote, bzw. sollten Sie per VNP oder Teamviewer zugreifen und darauf arbeiten, bzw. sich auskennen mit Hylafax und T38Modem. Ich versende die Faxe mit einem VoIP Account bei SIPGATE und SIP über einen SIP Server, sowie mit der command-line per sendfax 012344535234 beispielsweise dann, wie unter Linux.

We need to build a WebRTC platform that can bridge H323/SIP videoconferencing systems. Must work either on premise or as SAAS and allow integration to other SW platfforms. Development in Angular is prefered.

O projeto consiste em um balanceador de tráfego SIP. Os destinos devem ser recuperados de uma tabela de banco de dados MySQL. Uma ação (executar uma URL ou update no banco) deverá ocorrer se algum dos destinos retornar um código específico (Ex: 602). Não será necessário verificações de segurança (acl, etc) pois o ambiente de produção será rede local. Não será necessário adicionar serviços de RTP pois os áudios serão fechados direto para o IP de destino, resultado do balanceamento. Por ser parte integrante de um projeto, não será necessário nenhum tipo de interface do usuário. Preferência por instala&cce...

sip to whatsap, viber, telegram, signal getway

Applicable for Android App (reactnative, mongodb, nodejs) and website: 1. Hiding network details and just showing and describing network strength Excellent, Very Good, Good, Poor etc. But it should not affect other options anywhere. 2. Customising color Jitsi Option Menu as per our chosen design/color 3. Meeting Creation Link ...Server hosting independent and customised instance of Jitsi 4. Random name creator 5. Replacing Jitsi name with ouor brand everywhere in platform. For example if using our meeting from Muscat, someone got Jitsi error. So all error codes to be replaced with our error codes/numbers and explanations. Depending on if above is done in agreed cost, would consider for autoscaling, load balancing, Jigasi+SIP+VOIP, Recording upload to a link etc. I am in Kolka...

Hello, I am looking for an expert who can help me with sending VOIP pushnotifications for iOS (iPhone) devices using Asterisk / FreePBX environment. I am using Linphone SIP mobile client and I need you to help me configure VOIP Push service for the incoming calls. If you have experience in what I am talking here, then please bid on the project. Bid on this project only if you have experience.

We are a telecommunication company in Turkey which mainly operates in cloud pbx area. We need a softphone that will work with our pbx systems. It needs to work on 3 platforms; IOS, Android, Windows. You can see the detailed information below; • “Sip Server – Username / Password” based login screen - (We require this for the current project. After this project we are going to need default registration method for international use) • QR Code – We need to implement QR code for easy registration and configuration steps • Config file or URL – Same logic as QR. We could use this on computers. • Video Call Support – (H264) • Admin Panel – We require this for all platforms. It will show usage statistics and we must able to ...

We want to use Asterisk PBX with Avaya AAEP. The idea is to transfer the call when it reaches the IVR to another PBX (Asterisk) installed locally. Requirements - * Integrate Asterisk PBX with AAEP for SIP transfer * Call transfer can be both bridge or blind. Both, from Avaya to Asterisk and back should be supported * During transfer metadata like caller_number, call_language, etc should be passed

Looking for a PHP developer with the following skills to join team. ** NO AGENCIES PLEASE ** - Full understanding of MVC Frameworks like: CakePHP, Laveral, CI - Understand of REST API - Twilio SDK - Telesystems like SIP, SMS - Ability to work hours (600 - 1400) US Eastern Time. - Team Player. Able to take direction from Project Lead. - Able to communicate in SPOKEN English. - Good Customer Service skills. First 2 weeks will work as trial @ $10/hr. Negotiable there after.

Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

Need someone with specific experience configuring Ricoh Copiers for IP Fax using SIP.

I run painting parties on the weekend. Virtual Sip and Paint ;) But I have two full-time jobs during the week - and a family. I love painting and instructing paintings - but I need to promote the events two weeks in advance - so I need to have a painting composition two weeks early - but I am already so pressed for time. So I need a creative hand. A ghost-painter... y'know like a ghost-writer but for paintings. These can be made digitally or traditionally. ...THEY MUST BE YOUR OWN UNIQUE IDEAS (not from Pinterest. smh)!... BUT keep in mind, the classes take place over 2.5 hours and use 16x20 canvas and acrylic paint, so the composition must be feasibly accomplished by a lay-person (non-artist) with acrylics in that amount of time. It should be something you could easily...

I need an Android application creating which can act as a GSM gateway and SMS gateway. It needs to: Voice gateway: 1. Connect via SIP (or WebRTC) to my voice switch, which is hosted on the internet and not local to the phone 2. Accept calls from the voice switch server and place call out over the GSM connection. 3. Provide two way audio for the caller/callee Call generator: 1. Accept commands to place call on GSM side and make call 2. No need to connect audio to voice server 3. Play recording (or make DTMF noises) from phone over GSM call 4. Hang up call when required SMS gateway: 1. Connect via smpp to my SMS server (or use HTTP polling to a HTTP endpoint) 2. Accept SMS from my SMS server via SMPP or HTTP polling and send it out over the GSM network 3. Accept SMS from the GSM s...

...DRINK CUP AND FOOD HOLDER IN ONE AND THE MAIN PRODUCT IS FLAVORED FRENCH FRIES. AS A TWIST, WE PAIR IT WITH A DRINK LIKE SOFTDRINKS OR A FRUIT JUICE. THE IDEA IS CONVENIENCE AND MOBILITY WHILE EATING YOUR FAVORITE SNACK BECAUSE OF THE IDEA OF HAVING A DRINK AND A FRENCH FRIES IN JUST ONE PACKAGING. WE WANT YOU TO CONSIDER TRYING TO DESIGN THE BRAND NAME WHICH IS SNACK & SIP WHICH EMPHASIZES THE LETTER “I” IN THE WORD SIP AS THE STRAW. BUT IF YOU HAVE MORE AND BETTER NAME THAT SUITS WITH THE CONCEPT THAT WOULD BE GREAT! WHAT WE NEED IS SOMEONE THAT CAN CREATE A FUN, YET VERY RELEVANT LOGO FOR OUR BRAND OR NEW BRAND NAME OF YOUR IDEA. FOR THE LOGO CONCEPT, WHAT WE WANT IS FOR IT TO BE PROFESSIONALLY LOOKING BUT NOT BORING. WE WANT IT TO BE CARTOONISH, WITH...

I need to customize a Java open source sip auto dialer. The person needs to be an expert with a clear knowledge about voip. the opensource with the source is at GitHub: Basically, I want the software to perform a call hold, dial on the same port another call, then upon the success of that call, bridge that call with the call in hold. The software will dial to an IVR and the purpose of this job is to be able to use the same port 6 of a maximum of 6 calls, that is allowed by the sim card into the gsm gateway. so, we need to add that feature to open source. If you feel you can do this job, please send me feedback to discuss price and timing, we need this asap, thanks!

we need an expert in linux pulse audio or jack tools to configure a virtual audio between audio incoming from usb to a sip client in the linux

Having around 5.2 years hands on experience on Avaya servers like CM,CMS,AES,Mediagateways etc

Having around 5.2 years hands on experience on Avaya servers like CM,CMS,AES,Mediagateways etc

I'm in a need to setup a SIP trunk that I have into my Voice Cisco Router 2901. The Voice Router is behind a Cisco ASA device. I do have access to the ASA device to setup NAT or any other configuration needed. Thanks

Hello I install a FreePBX with Custom Destination. It is connecting to my Provider using 7 IPs for SIP any there many IP for RTPs. The SIP Trunks are UPnbut when I call a DID, the call reach the FreePBX but there is no voice. I need someone help to trroubleshoot that. Thanks

To set up installed Asterisk PBX stable 16 qnap x86 SSH small set up 4 sip 5 phones 5 softphones GUI - web admin simple voice missed number to email debatable option

We are looking for a custom-built SIP Softphone (PJSip Asterisk extensions) project using .Net 5 and C# to with the following features: 1) Source Code in Visual Studio project using .Net 5 for Windows Application 2) Downloadable internal directory/contacts 3) Simple UI 4) Minimizable to Tray 5) DTMF and Multi-line with hold, conference, transfer 6) Configurable ring notifications 7) Cannot use proprietary/paid license components 8) Completely configurable from web endpoint (can be XML/YAML/text for the extension, server, etc) 9) Code standardly documented in English

Looking for art instructor with personality, group training experience, experience with acrylic paint - water based paint and canvas. Must be able to work with beginners in a creative environment with a smile. Opportunity to present art portfolio in the artsy village of Nyack NY. Budget is open.

Hello, Few call center employees need to work from home, so VPN need to be set up to connect them from home to head office network to make & receive phone calls. Input data: - We use PRI line with 100 numbers, out of each around 5 need to work remotely. - PBX is Asterisk-based; also Yealink, windows server 2012, Call center agents use X-lite old version as free SIP-phone tool. If needed, there is DyDNS address of the head office, as head office IP address is dynamic. Server & PABX map is attached. Time to finish project : 2 days + 1 day testing. There is no VPN existing in head office currently. Call center laptops at home for connection with VPN have Windows.

I have two Avaya IP500 V2 PABX systems. I have a primary site that is working without issues. I have a second site that is connected via VPN with a second unused Avaya system. I want to be able to connect the two systems together so that both sites act like one site. I need to be able to take calls at either location and forward calls between the the sites. This can be done remotely using Anydesk or teamviewer. my computer has access to the web configurators for both ends. Thanks.

I have goip-1 Firmware Version: GHSFVT-1.1-67-4 on my local network. it is connected as dmz and I can give the web interface for configuration. My freepbx installation is hosted on Digital Ocean droplet. Sim is installed in goip and calling is active, just need to configure the trunk, inbound and outbound route. Please only bid on the project if you know freepbx and goip gsm gateway

I need an iPhone app. I already have a de...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

...strategy of competitive brand`s speciality teas in the e- commerce / online platforms. Also need to understand the customer preference for the type of tea and flavours 1. Details on Market Overview- Trends, Drivers, Challenges for this market. 2. Current market size estimation and forecast - till 2025. 3. Overview of key players – ( Organic India, Twinings, TGL, Vadham, Tea Box, Tea Monk, Tea Trunk, Pukka etc.) • Sales and Market Share • Product Type • Price segment & Pack format • Core competencies / Offerings. Adopted marketing strategies of the different brands etc. 4. Region wise Market Share ( Delhi, Calcutta, Mumbai, Chennai , Pune, Hyderabad, Bangalore, Cochin etc) 5. Ecommerce /Online Channel wise Share. • Sales of Special...

Hello, the project is based in develope a configuration into a CISCO ASR1002 to customize a SIP/SS7 signiling switch. The SS7 signiling is received in TDM/E1 using a Channelized STM1 Interface connected directly to the Operator, inside the STM1 the first 8 E1's has the SS7 link & CIC's. The system will have to convert to an ITP / Point Code for SS7 signaling as a part of the final solution to be used as a SIP SBC or Sigtran point. Components: 1x CISCO ASR1002 1x Channelized STM-1 port (Cisco 1-Port Channelized OC-3/STM-1 Shared Port Adapter), the stm1 has 8 E1's. 2x SS7 Link (OPC/DCP/ADJ Point)

...certainly be a plus, but is not a requirement. The type of jobs that we'd be expecting the freelancer to perform are: 1. Listen through the un-edited version of each new episode and annotate time-stamps for editing (e.g. delete section from 1:39 to 1:52). The positions for cuts will be obvious, as it is usually sections where the podcast host and guest discuss logistics, take a break to have a sip of coffee etc). Each episode will be roughly 60 min long. 2. Annotate the timestamps where images would be added (we'll provide the images, so it would just be necessary to find the right timestamp when that image should be shown in the video) 3. Annotate timestamps for sections that are "highlights". Those will be cut together separately for a shortened summary/h...

El sistema debe estar en php/bootstrap la idea es poder ingresar usuario sip (voip) en una plataforma web y que automaticamente se creen en asterisk (estoy usando VitalPBX). Debe tener: Creacion de usuario Modificacion de usuarios Eliminacion de usuarios lista de usuarios activos y no activos eso para comenzar luego podemos ir añadiendo mas funcionalidades

Searching for the valuable software house who has exp on developing iOS and Android Mobile application on SIP protocols with the help of Admin panel and Must have experience on asterisk or elastix. Deep Project details will be provided over the PM.

The app is 2 pages (+ login page) Page o...Remarks: 1- There will not be much work on the dialer page as we can implement a ready open source project. check (Linphone) 2- login will be optional to the user to login using Facebook/ gmail or to skip In both ways the app will be fully functioning and storing the credit locally 3- We will implement admob + Facebook ads alternative to each other’s 4- Connection to the voip server will be don’t SIP direct connection (no API needed) and for security we will keep the authentication details (IP,username,password) on the database (firebase) 5- the connection details to the server ( IP/ username/ password ) and the cost per min are to be stored on a server ( firebase ) 6- cost per min is to be a variant that I can change later ...

I would like to get the SIP trunk configuration for goTo and Nextiva sip provider