Pstn jobs
I'...is voice calling. Moreover, it's critical to have the ability to call PSTN (Public Switched Telephone Network) to WhatsApp. The ideal freelancer should: - Have previous experience in cross-platform app development, specifically with VoIP integrations. - Familiarity with APIs and how to integrate them into both iOS and Android platforms. - Familiarity with WhatsApp API would be advantageous. - Understanding of the requirements and constraints of the PSTN. - Necessary skills in developing voice calling functionality. This project's success will hinge on your ability to create a seamless experience for users to engage in voice calling from VoIP to WhatsApp across both iOS and Android platforms, including calling from PSTN to WhatsApp. Ready to ma...
I'm in need of a skilled professional to integrate direct routing within our Microsoft Teams setup using an AudioCodes Session Border Controller (SBC). Our organization runs on a hybrid PABX environment, and we're looking to enhance our system's functionality and security. **Requirements:** - Configure AudioCodes SBC for direct routing in MS Teams - Connect external PSTN calling to MS Teams - Enable extension dialing within our organization - Facilitate call transfer capabilities between Teams and our hybrid PABX - Implement IP-based access restrictions - Ensure compatibility with our existing security framework **Ideal Skills:** - Proficient with Microsoft Teams and AudioCodes SBC configuration - Experienced in hybrid PABX systems - Knowledgeable in telecommunicati...
Fusion PBX installation integrated with PSTN SIP trunk, Webphone(SIPML5 or any other for Audio, Video and conference), Webchat and Screensharing configured with Webrtc enabled. on GCP ubuntu22VM Recording Enabled
I have a server with an FXO card, the scenario that i need to do it is this: 1.- i have extension 100 with a mobile phone in a follow me destination 2.- if the mobile phone it is not picked up in 9 or 10 seconds needs to send to voicemail or IVR or another destination (like a ring group) the main problem is that i cannot make the asterisk to understand when the PSTN call exceeds that ring time because for asterisk the call it is answered so i need a help to setup correctly my box for make it to work please only experienced freepbx / asterisk users needed
I am looking for a skilled app developer to create an app that allows me to receive GSM ...knowledge of internet connectivity and remote communication protocols. Basic requirements for the project with example: For example if the SIM card with mobile number 9999999999 is inserted in Phone A (Android xyz) device and if any xyz person calls the number 9999999999, the incoming calls should be received via internet or any mobile app like Zoiper etc or SIP server. Basically it's conversion of PSTN to VOIP or SIP. Via internet connectivity, the user should remotely be able to seamlessly answer all his incoming calls landing to his normal SIM card which may be functional at a different location. If you have the necessary skills and experience, I would love to discuss this project fu...
...and immediately notify NTT Global Networks Team of any damages. 2. The engineer shouldn’t start any installation work without obtaining a go ahead from the coordinating NTT Global Networks representative. 3. The engineer has to build a connection between the provider router/NTU and the NTT Global Networks router, as instructed on the phone. 4. Engineer needs to connect the customer provided PSTN line to OOB modem and that modem should be connected to console of the router. 5. Proper routing of cables and labelling equipment/cables. 6. NTT Global Networks Engineer will be on call with your engineer and he will help your engineer to complete the activity. 7. Engineer to share post install pics, Site Visit Report with NTT Global Networks after completion of the activity...
I am looking for someone to setup and configure Janus Gateway with SIP video for a 1-1 caller. The caller will be using SIP video, while the callee will be using PSTN/SIP audio. I have no experience with Janus Gateway, but I prefer Node.js for this project. We will be using an Other SIP provider. SIP video callee side will see video but no need to stream their video . Need to setup Video conference like (jitsiMeet) also Skills and Experience: - Experience with Janus Gateway installation and configuration - Expertise in Node.js programming language - Knowledge of SIP video and PSTN/SIP audio - Familiarity with Other SIP providers
Job Description: Have installed the latest version of FreePBX on new hardware and Cisco SPA8000 . Requiring a talented Freelancer for this system to: Connect to Yealink T28P x 8 Connect to 2 physical PSTN phone lines 1 Fax line via Cisco SPA8000 Route calls from Cisco SPA8000 to FreePBX A separate inbound queue for each phone line Outbound line selection based on grouping Configure outbound email notification! Headset programming Set up / Configure Sangoma Softphone App Any other minor tweaks. Immediate start 5/1/2023 10:00am EST! Skills: Asterisk PBX, Linux, PHP, VoIP
...spam (contact page, subscription page and other). /add /disclaimer-adult I have nothing, it's up to you ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;streaming chek best free solution ????? WebRTC Hosting For live chats to work you will need a Windows based VPS or a dedicated server running the RTC libraries. You can also opt for RTC hosting. In almost all cases, PSTN hosting is the cheapest solution. Hosting STUN/TURN You will need a VPS running a STUN/TURN setup to ensure all connections go through. STUN/TURN hosting is a cheaper alternative that you might want to consider. ................................................................................................................. domain: done hosting: done: server centos7: and pan...
I would like to configure a scalable VoIP server hosting on-premise and integrate it with Exotel for PSTN dialing.
...Sip and webrtc integration. • VOIP software development. • Good Knowledge in PBX, SIP, RTP protocols. • Worked on Queue, IVR and Voicemail related applications. • Expert in Freeswitch installation, configuration and... • Competent enough to setup daily call limit and concurrent calls Requirements · Software Development experience in Freeswitch, FusionPBX, Opensips, SIP, VOIP, SDP, TDM, IMS, PSTN, Python, Perl, Linux, and Open Source Technologies. · Strong Technical, Logical and Debugging skills with innovative and result-oriented approach ·working experience in Python, Shell, Perl, Asterisk, Freeswitch, Opensips, Kamailio VOIP, SIP, IMS, NGN, ISDN, TDM, and Telecom/Network Protocols. · Very Good Knowledge of VOIP/SIP server...
HI we need a script to be run on Ubuntu Server ( text based environment ) and do the following automatically : 1) sign into skype account with a username and password 2) edit setting in the call forwarding features ( which was set previously ) , this is the URL : :18888005555 each time , the number will be changed, 4) save edited number, it can be any scripts in PHP, Pyton .... Regards, John
We need when we setup did action call to pstn callerd id should be the of the caller in forwarding.
We are looking for a simple Twilio app that will prompt a caller to record a message, and then call out to another telephone number and replay the recorded message on repeat for some period of time (i.e. 3 minutes). This app just needs to be responsible for answering the call and then placing the outbound call via either PSTN Call or Twilio Elastic SIP Trunk. If we need to run code outside of Twilio it will need to be in a Docker container so Dockerfile will need to be included with the code.
you have to install freeswitch with astpp billing panel. from Astpp all uer & DID, trunk will manage. Voip call should work between PC to PC & PC to PSTN.
We have an iOS app, just changed our server, we are facing a problem calling to PSTN number, if you have experience in developing twillo call recorder app or can do this task, you can apply.
Configuration of DVG600s to send calls, and configure cloudfonica pbx to send calls to the gateway the gateway will send the call to PSTN. only 1 channel FXO. Please consider this requeriments: Cloudfonica PBX should send SIP Invite packets to Dlink Gateway. If FXO lines form PSTN are not working, calls can not go through.
i need to configure D-Link DVG-6004S fxo to be able to dial out calls received from sip through pstn lines
Looking for A2billing installation on Centos 8 with latest stable Asterisk Version. Features : - 1) PC - PC 2) PC- PSTN Calls 3) Callback 4) Inbound DID 5) A2billing integration with freepbx 6) A2billing integration with Vtiger
A2billing installation on centos 8 with latest Stable asterisk. Feature : PC-PC Calls PC-PSTN Calls ANI Callback Web Calls Back Inbound DID A2billing integration with Vtiger CRM
We are looking for a simple Kamailio configuration to act as a SBC of sorts. We have a SIP based fax server that isn't able to fail-over to the next peer if the call fails. We would like to have Kamailio try a list of servers sequentially until the call completes. We only need Kamailio to handle egress calls from the fax server to pstn. We will additionally need to proxy media and will require direction on configuration of this aspect as well. Solution needs to work with t38.
We are looking for a freelancer to add inbound PSTN call support. Outbound PSTN call support is already implemented including call transfer and call conference. Please review the attached sheet for requirement. Please let me know the timeline and quote.
Step 1 :- A server ------> Invite --->> Bserver ------->> Invite ------->> PSTN Step 2:- A server <------ 100 trying <<--- Bserver <<------- 100 trying <<------- PSTN Step 3:- A server <------ 183 session Progress <<--- Bserver <<------- 183 session Progress <<------- PSTN Step 4:- A server <------ 183 session Progress <<--- Bserver <<------- 180 Ringing <<------- PSTN A Server >>> Installed Asterisk with Freepbx B Server >>> Installed Asterisk with Freepbx PSTN (SIP) attached with Server B An extension 123 created on Server B On Server A created a SIP Trunk with configuration of extension 123 that was created on Server B Also co...
I need to setup a PBX for businesses I operate with standard features, call recording, call routing using PSTN lines. One of the modules I am seeking is API support in the solution deployed.
Hi Nsikak A., I saw you already have a click2call developed for a client, my platform is ASTPP 5.0 with freeswitch. I need the customer to have the option to choose to receive the call on the extension or pstn, and after answering the freeswitch sends the call to the site visitor. Whenever both incoming and outgoing calls are PSTN, they are billed in astpp. It can be in the two scenarios you already have: 1. Click to dial by users 2. Click to call Thanks
Configure new account install PSTN configuration Configure 2 physical grandstream devices
To develop the following in bigbluebutton 1 Vanity URL: Develop a scheme where Home Room URL is 2 Download recording: Recording to be downloadable in MP4 format 3 Spotlight on Speaker: When speaker speaks, his or her face should be highlighted or should occupy a larger screen space 4 Reposition Leave meeting icon: Leave meeting icon to be placed o...URL: Develop a scheme where Home Room URL is 2 Download recording: Recording to be downloadable in MP4 format 3 Spotlight on Speaker: When speaker speaks, his or her face should be highlighted or should occupy a larger screen space 4 Reposition Leave meeting icon: Leave meeting icon to be placed on the main screen ( next to mute/ unmute key) . 5 Meeting invite to have PSTN dial-in information
I'm looking for a Power BI expert who can build Power Query to hook to our current database and be able to provide real-time information based on certain criteria. Deliverable: Develop a query and report Specifics: 1) Currently using the CQ and AA Combined An...multiple choice filter options 2) Connect to our Noam database on the Microsoft platform for real time data 3) We want to filter options for date range, call type, and user display name. We want to select one or all choices in the input filter. 4) All dates should be converted to the eastern time zone (currently using settings on the desktop - DAX input) we recently switched to teams voice as our PSTN provided so I'm very limited on Power BI. Very open to your suggestions and thoughts! I look forward to hearing ...
I have Ten years experience as an ICT/Telecom Engineer. My job description includes Ensuring that specification and requirements of all ICT/Telecom projects are fully achieved and executed according to the timing, PSTN, VoIP Protocols and Translations of core switching technologies ,engineering parameters and peripherals, Industry standards and best practices, and change management principles, Deep knowledge of IPV4, IPV6, Fiber Optic Technology standard, requirement, system analysis, design, installation, procurement of fiber optica connections, Provide design calculation and drawings for construction of telecom projects involving microwave communication, channel banks, SONET multiplexer, teleprotection, fiber optic networks, C band/Ku band satellite, Tetra/VHF/UHF mobile radio, Po...
Implement asterisk solution - Asterisk Dial-plan, Asterisk Gateway Interface and Asterisk Manager Interface. - Configure Queue, IVR and Voicemail -Asterisk installation, configuration and troubleshooting. - installing and configuring Asterisk with PSTN card.
...and serverless architectures Preferred Qualifications: Previous experience working with distributed architectures, microservices, API's, telecom infrastructure, or developing and deploying production level machine learning/AI products is an asset. Experience with DevOps, CI/CD, security protocols, & quality control/automation. Experience working with telecom infrastructure (e.g. Twilio, PBX, PSTN). Bonus Skills: Experience working with common data science toolkits, such as NLTK, OpenCV, NumPy, etc. Experience working with ML techniques and algorithms, such as CNN, RandomForest, Naive Bayes, clustering etc. is an asset. Experience working with OCR or computer vision-based document processing (such as Tesseract, Google vision, Azure vision, ...
I have freepbx installed on my server here at the building I have a public IP address that allows me to access the freepbx GUI from outside the network am using (Ubiquiti network equipment ) I have GSM Gateway (Dinster2000) linked to the network also can access it from outside the network I have PSTN Gateway (Grandstream GXW410X) linked to the network also can access it from outside the network I have PCI analog telephony (Sangoma) the PCI chip is placed in the PC server of freepbx I have IP Phones (Fanvil 600 + Cisco Cp8865) linked to the same network I want to configure all of these devices and make it work like the attached flow also I want to be able to link to the freepbx using softphone outside the network (note some of required tasks are already works but I need you to m...
We need a PBX on a Virtual Machine based on Asterisk (freepbx or whatever you think is best). - Create 5 extensions. - Configure sip trunk (with linksys spa300). outgoing pstn / incoming connections . - Configure a GOIP Gateway (1SIM - GOIP) for outgoing / incoming calls. - Configure outgoing calls with international prefixes. - Record calls. - Enable video connection with h264. Thanks,
...and serverless architectures Preferred Qualifications: Previous experience working with distributed architectures, microservices, API's, telecom infrastructure, or developing and deploying production level machine learning/AI products is an asset. Experience with DevOps, CI/CD, security protocols, & quality control/automation. Experience working with telecom infrastructure (e.g. Twilio, PBX, PSTN). Bonus Skills: Experience working with common data science toolkits, such as NLTK, OpenCV, NumPy, etc. Experience working with ML techniques and algorithms, such as CNN, RandomForest, Naive Bayes, clustering etc. is an asset. Experience working with OCR or computer vision-based document processing (such as Tesseract, Google vision, Azure vision, ...
install gui configure wan internet port 2 vlan- phone - pc pc connection from phone 2 fxo group call forward call contact list fxo0 hunting group all 201,202,203,204 at same time fxo1 hunting group all 301, 302, at same time extension: 200 201 202 203 204 301 302 calls out from 200 to 204 to fxo1 call out from 301 to 302 to fxo2 pattern output call to pstn - 9 digit direct to pstn - 10 digit starting with 6 direct to pstn
What I want to achieve, I can describe as follows. 1. I have a valid PSTN phone number. 2. If someone calls my number through PSTN, my analog phone connected to SPA1001 shall ring and I can make a conversation 3. I am able to call someone through PSTN with my analog phone connected to SPA1001 and I can make a converdation I think this setup wouldn't be a problem for such an expert as you. I have computer skills (setting up LAN networks and so on...), but I am not an expert in the IP telephony field. I have connected all the gear, I can manage SPA3102 and SPA1001 through their web interfaces. Booth SPA devices have fixed IP addresses.
I want to develop a web-based application where the frontend has the following: 1. Users login to the web site using Single Sign-On 2. Once the user logs in, they can use AWS Chime to record video and store video in S3 3. User can click past recordings and view the video 4. All of the user's data and metadata stored in DynamoDB 5. Users can pay for su...where the frontend has the following: 1. Users login to the web site using Single Sign-On 2. Once the user logs in, they can use AWS Chime to record video and store video in S3 3. User can click past recordings and view the video 4. All of the user's data and metadata stored in DynamoDB 5. Users can pay for subscription using credit card See the attached image. Most of the architecture (without PSTN meeting service, Am...
We have a single channel SIP account with Skype Connect (Skype Manager) as a trial, more channels will be added. We want to integrate to our asterisk server to make outgoing calls via Skype. Please, we only interested those freelancer who have done it before and not those wanting to experiment. Project is only consider done when we can make calls from SIP phone to PSTN number via Skype Connect Thanks
Need to upgrade existing Free PBX platform and insure redirection of its press options to connect to a sip server via webphone/softphone. Experience with call routing on selections from PBX options to sip server and interconnectivity thru voip(softphone) and pstn are required. Knowledge of Javascript and API implementations(Documentation is available) are required to hand-shake Mizutech webphone software with the FreePBX. Migration of files from external hosted server to the PBX server may be required.
...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 quick job. NOTE: If you need to ask any questions before bidding, please do so. Please do NOT place a "placeholder" bid then tell us t...
I have an iOS and Android app that allows users to set up a custom voicemail service. To enable the voicemail service, we identify the user's phone carrier (Verizion, T-Mobile, AT&T, etc.) and have the user se...and then look for an incoming forwarded call from that user's phone number. However, it seems like there could be a simpler process. The second option I am trying to figure out is a way that I can just call the user (it's okay if the user's phone rings once or twice), and then retrieve the actual SIP header information and extract the call forwarding phone number. This way we can see at the SIP / PSTN level the number that is configured for the conditional call forwarding. To apply let me know if you know how to do this, and how long it would take y...
We have Webex Calling service and want to connect local gateway router to Webex for PSTN lines.
Hello all, I am looking to configure a Voip to PSTN/ GSM connection to several countries in the Middle East and Asia. What I basically need is to have an app on my desktop (or a web interface) that will allow me to call my clients and business partners. few basic requirements are: 1. connect me with a GSM provider (from my PC through the Voip network and the GSM one) 2.a local number as my caller ID (to be seen by the second party) I intend to use this line for the longer run so I will need support for the system throughout the operation. Please contact me if you find these requirement within your field of expertise. Thank you, Benny
...between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsite APP communications. Establish a webRTC to SIP(PSTN) connection having two-way audio call and local VoIP connections (Guard Phone). If an alternate viable method is proposed we can consider it. The POC must include a means of adjusting resolution, i.e. VGA, and framerate, i.e. 10 frames/sec. The company accepting this project will have knowledge of codecs and echo cancellation. We are looking for a freelancer or age...
I need to make Voice over IP calls between 2 countries, to landlines and mobile numbers: In both countries I ALREADY have "landline" service (PSTN) with unlimited call plans. I would prefer NOT To use an Asterix, Elastix or similar solution, in that case I rather to use a low cost solution such as the old Linksys SPA3102 or similar device (with FXS, FXO ports) I know that there are “virtual PBX” services with low costs BUT if I already pay for unlimited calls service, and I DO NOT want to pay additional monthly costs or additional costs per call. So, What What I need ? : I need an explicit diagram of a low cost solution using my current PSTN lines . Preferably be based on ATA-type devices like the Linksys SPA 3102 or something similar cost sol...
We are using Asterisk 13.32.0 with FreePBX at the moment and are in need of a SMS feature for customers to text to the PSTN along with receiving texts. please PM for further details. Also there could be lots more work available depending on the level of professionalism and quality we see from this task.
I require a SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and freepbx. I need a SIP & Jitsi expert for: 1) create a sip/h323 connector for jitsi, 2) help me with getting an affordable india phone number for inbound connections – Twilio pricing is not affordable for Indian numbers 3) setting of sip trunk between jitsi and freepbx, so the participants from pstn can connect to the jitsi instance 4) Connect our jitsi instance to google transcription. I will provide the API key
The project is to create SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and fusionpbx. Need a SIP & Jitsi expert to: 1) create a sip/h323 connector for jitsi, so, it can be given to hosts. 2) setting of sip trunk between jitsi and fusionpbx, so the participants from pstn can connect with jitsi instance deployed in our cloud.
I am looking for help in configuring a Mediant SBC for multi-tenant use. We want to have a shared SRD for the PSTN side and Isolated SRDs for customer trunks. For the PSTN side incoming calls we are hoping to use different ports to be able to classify which customer to route to - i.e. port 5100 will route to customer A, port 5101 to customer B, etc.
