Sip trunk elastix nec sv8100trabajos

...dialer, power dialer, and voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We will provide the sip trucking and server. Here is the basic requirement: Looking for a call center software developer 1. Able to devel a power dialer, Predictive dialer, voice broadcasting, VR, survey campaign in a single platform. 2. Good knowledge of SMPP Integration 3 Good knowledse of Asterisk AМI voice broadcasting. You can't use any existing company templates due to соpyrights. It will be a 20 sip trunking асcount. If you have the experience, let us know. We vwill provide the sip trucking and server. Here is the basic requirement: Looking for a call cente...

We have an environment with VMware vSphere 7.0 and 4 hosts ESXI. Each esxi have a pnic connected on ubiquiti edgeswitch 16-XG (10gb port) In VMware we have a distributed switch configured and a two different distributed port group with VLAN 70 and VLAN 5. On the first host w...windows VM on the IP: and on the second host we have another windows VM with the IP: 192.168.70.3. Both VM are attached to portgroup vlan 70. These VMs do not ping eachself. If we remove the vlan id on the portgroup and on the physical switch we can ping. The problem is identified on configuration of ubiquiti switch. We require an expert on ubiquiti. We have problem setting the trunk port on ubiquiti. We will pay only if the problem will be solved. Please answer only if you can solve the problem.

Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wakeup and receive call. We can make this flow work with some device but not all device work smooth, some device can not wakeup ( android, ios too) i think the app like that ( voip ) need some special skill and tech to implement. My requirement: please see the attact file

i want someone to configure 20 Channel SIP Trunk in my 3cx Server

web base sip dialer i need to develop

I have a few extensions behind a pfsense firewall that can’t connect to my hosted FreePBX. I need help configuring the firewall for this to work.

I am starting a mens swim trunks company called Trunkaholic, and I would like the logo to say Trunkaholic, with a cartoon elephant or an elephant trunk incorporated within the logo, also maybe some waves or beach stuff. My bathing suits for men will be mainly focused on animals incorporated in the bathing suit, like penguins or flamingos. I would like the logo to have a tropical vibe, and think modern, rather than cartoonish. My first bathing suit that I will be releasing will be linked down below. PLEASE MAKE AN ORIGINAL LOGO OR ELSE YOU WILL BE REPORTED.

Looking for Engineer to help configure SIP trunking to ITSP and CUBE

Mobile softphone dialer for android and IOS. Voip enabled using our sip server. You will need our SDK for android and Java to develop it fully. it simple and straight forward.

we need someone who knows FREEPBX and can login via ssh to see why calls are not working via out trunk and route

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I have FusionPBX which works on Freeswitch. I have discovered a bug in the system. 1) When normal external call is made from an extension number in fusionpbx, the sip header contains the extension number and DID number. This is fine. 2) When call forwarding is activated the extension number and DID number are no longer in the sip header. This means that the call cannot be authenticated at extension level when sent to the carrier. I need this to be fixed so that the as a bear minimum the extension number is present in the sip header when call forwarding is activated. Example is attached

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 ...

... Ec/Io, RSCP, RSSI, PSC, RRC State,TX Level, Neighbor Cell Measurements, HSPA+ DC in use, Audio Codec Type, AMR Bitrate - 2G/GSM: Band, RxLev, RxQual, C/I, TxPower, ARFCN, RRC State,Serving Cell measurements, Neighbor Cell Measurements, Audio Codec Type, AMR Bitrate - 4G/LTE: Band, RSRP, SINR, RSRQ, RSSI, EARFCN, RRC State, - 5G NR: Band, SS-RSRP, SS-RSRQ, SS-SINR, BeamIndex, SS-PCI - VoLTE/RTP/SIP Layer - Lock NV Item Chipset based api - Layer-3 Messages - GSM RR, WCDMA RRC, CC, MM must be captured and written to a plain text file. The programm must run on Qualcomm based phones. Requirements for participating: - Understanding of GSM, UMTS, LTE & 5G NR technologies - Understanding and experience of Layer 3 tracing and decoding - Experience with Qualcomm chipsets - Providing a...

I have a 20channel SIP TRUNK from Tata.. I want to configuration it in my 3cx system.... and aalso enable API of 3cx for my crm

i need to translate response codes from asterisk termination to origination for example i am getting sip response codes 480 i need to give origination 503

I need to install a stable version of Opensips 3 with a web interface on a server running Centos 7. It will be used as an SBC to hide the network topology, change sip headers and load balancing.

We are a non profit named Share & Care Releaf! The misspelling of relief is intentional. We are a charity who's primary focus is the care, beautification and maintenance of trees. We will assist those in need of major expensive tree work that effects life and property. We help in pubic areas like parks that do not have proper tree management. We will assist in national disaster eve...property. We help in pubic areas like parks that do not have proper tree management. We will assist in national disaster event that involve emergency tree work. Please make the letter A in Releaf an actual leaf. We like the heart and tree combination. Releaf should be spelled with an actual leaf in place of the A. We would like a real tree rather than a tree with a representation of a person a...

Need an expert to install a phone extension system. I have an office but only one receptionist works there with one phone line. I have more business and I would like to use each business with one individual extension number like e.g. accounting 1234512345 / 1001 ext. and monitoring 1234512345 /1002 ext. If somebody call us, the system picks up the phone and request extention number. Based on given ext. number by caller the phone display shows the pre-programmed business name to that ext. number and receptionist can see what business is called (e.g. "accounting" call). Task: advise what system I have to buy and how I have to program it. Thank you

I need to setup call center software and multi tenant fusion pbx as sip server with soft huawei HG8245 as client gateway

2 offices located 1 hour away from each other. Raspberry Pi4 with 3cx PBX at “main” location “Main” location has fiber optic internet 200/60 down/up speed, other location cable 150/50 down/up Public IP but not static Each location has Gateway VOIP Grandstream 4 Fxo GXW-4104 Each location has 2 landlines and 1 SIP/VOIP line Various phones to connect, most are Grandstream GXV3240, 3370 and 3275 No operator per se, small business where whoever is available takes calls Both locations have battery backup on equipment I want to connect all lines from both locations so people at 2 offices can call out, answer, intercom between phones. Ability to take a phone home and be connected to office for the casual employees or for covid lockdown. Make sure phones are pro...

We need a ringless voicemail system created for us. What's the best deal you can give us and what are the actual sending rates for it? Does SIP help to cut down the costs compared to using something like Telynx?

Set up CISCO UC set up in the developer sandbox at Hook it up to out SIP line via twillio and set up VXML and MRCP to communicate with a dialog engine. We will supply the ASR and TTS and the NLP end points.

Hello I would like to know if you can do the following, I have Jitsi Meet 2.0 currently in a vps in ovh working well and I just installed issabel 4 in another vps both are seen by local ip, I already have the annex, sip trunk, for integration with jigasi, and enable the audio transcription to be displayed as subtitles, I also have the dial plan, you can make it work, thanks

Hello I would like to know if you can do the following, I have Jitsi Meet 2.0 currently in a vps in ovh working well and I just installed issabel 4 in another vps both are seen by local ip, I already have the annex, sip trunk, for integration with jigasi, also the dial plan, you can make it work, thanks.

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.

...a patch to existing NEC V850 Firmware Expert will need to be able to : - Write in C/C++ that can create arrays/look up tabale - 2D and 3D Lookup tables (see example) - Look up table will be programmed in C/C++/ etc (see example image) - Write algorithms for calculations for these arrays, multipliers, dividers based on set values and necessary code - Write any other supporting code / calculations - Example code ideas will be provided. Example of code : If variable A = B > go to routine with lookup table C. Get result of C. Multiply result of C by D, place result (E) into RAM. REQUIRES: Ability to Compile the code into a BINARY FILE that is NEC V850/NEC V850E Processor instructions. - MUST BE ABLE TO COMPILE THE CODE (even if incomplete) INTO A BINARY FILE THA...

...a patch to existing NEC V850 Firmware Expert will need to be able to : - Write in C/C++ that can create arrays/look up tabale - 2D and 3D Lookup tables (see example) - Look up table will be programmed in C/C++/ etc (see example image) - Write algorithms for calculations for these arrays, multipliers, dividers based on set values and necessary code - Write any other supporting code / calculations - Example code ideas will be provided. Example of code : If variable A = B > go to routine with lookup table C. Get result of C. Multiply result of C by D, place result (E) into RAM. REQUIRES: Ability to Compile the code into a BINARY FILE that is NEC V850/NEC V850E Processor instructions. - MUST BE ABLE TO COMPILE THE CODE (even if incomplete) INTO A BINARY FILE THA...

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 ...