Openwrt asterisk projectроботи
Hello, we created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho do...created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho doesn t appear any pop up Will be provided a vpn to access Asterisk pbx and admin l...
We are looking to own an open source(preferably) SIP media server and stream audio on it. This has to receive audio stream from asterisk. This has to start automatically when conference starts up on asterisk (our pbx). We have the PBX'es, just need that audio stream / internet radio that can be accessed by those who can't call in.
hello I hope you all are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. Budget is tight as well as time schedule but there is a great opportunity also for the maintanance phase.
Project details The project is a videoconferencing system using PHP and jitsi. It will require development to use jitsi apps as native, or our own apps based on jitsi opensource code. We will leave the joice to the developer to advise. Background Zoom () is a full fledged video conferencing system built on top of jitsi video and offers meeting and chat, webinar, and conference room systems. Jitsi () is an opensource videoconferencing software designed to hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. https://zoom
I am looking who understand voip scanning . basically I am looking to install following program from start : please read and let me know if you can do this for me ?
we are interested in VOIP Asterisk PBX with ISP and IAX on Puppy Linux where Internet provder is blocking RTP/i/ports so need some encrption or tunneling also
Theory of operations: 1. Customer will dial a satellite nr 2. Asterisk will recognise the pattern and will add an additional string in front of the number
I am looking for some advice technical assist with an issue that we are having with our asterisk based PBX. We have been running an asterisk setup on our internal network for several years with almost no incident. We have two ISDN lines that connect through a B410 digium card to a wazo based asterisk box. Due to us now needing to work remotely we are running into issues. Everyone has their own physical SIP based phone at home that connect to our internal work network. This network has a public IP so there are no issues there. We have also set it up with the following IP rules: 5060:5160/udp ALLOW 10000:65535/udp ALLOW I have got the phones mostly working but I seem to be running into intermittent issues. These issues
I'm running a PBX (Private Branch Exchange), which should be accessable from the internet and from OpenVPN. The PBX is running in a datacenter, so there's no other NAT. OpenVPN's interface ist tun0 with Asterisk PBX is running on The "world" (the internet) is provided over eth0. OpenVPN is reachable from the internet on eth0 port UDP 443. The PBX should be reachable from the internet on port UDP 5060. Since some guys have NAT problems at home, the PBX should also be reachable from OpenVPN. I need someone, who designs the DNATing rules for me, perhaps a policy based ruleset is needed?
i want to implement webrtc on one of Customer development project I want to implement webrtc server software like Asterisk or FreePBX
We are looking for an asterisk / full stack developer who can help us build a custom dialer platform for call center using Asterisk as the backbone This project is a VoIP Platform for Agency Call Centers mainly for high volume lead call consumption across various types of health, insurance and mortgage products. Here is the basic outline of the project: This project will require the following: 500 outbound channels sip (spanability) Administrator GUI Agent Panel Phone list builder CRM - Client Manager with Followup Scheduler for Agents, commission tracking, retention Sales Reports Vendor Manager (Add, Modify, Delete Vendors) Lead List Dnc database and scrubbing tool HIPPA / PCI Compliant Statistics page .wav file uploader and player for outbound Emai...
I need someone to help setup an emergency call center using asterisk. The system should be able to handle 100 concurrent calls (server had been procured). We will start with 10 agents and they we would like to use a soft phone to connect the agents on their laptop. 1. Call Transfer, Hold, Forward 2. Call Recording and Playback 3. Configure IVR tree We will need the asterisk box configured with a local sip trunk provided by the telco. Soft recommendations welcome otherwise can purchase something like Please this is an urgent requirement and only have 3 days to deploy it. less
hello I hope you are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. I would like to hire you directly please contact me and disucss the project.
I am looking for some solution based on asterisk or freeswitch which provides me similar features as of multi tenant pbx like unlimited plan in please apply only if you can deliver, demo will be required.
I have Issabel CallCenter (elastix ) I need to add AMS (aswering machine detect) on this server Issabel working with Asterisk 11
Looking for someone who has experience with openwrt & mac. Trying to flash openwrt firmware to a tp-link archer c50 via tftp. Please don't bid unless you have direct experience with tftp, openwrt, etc. I need someone who can instruct me on how to install and setup the firmware (via live chat). Thanks!
Hi, I have a SIP Trunk from Saudi Telecom Company (STC) in Saudi Arabia and I want to configure inbound and outbound calls. The SIP trunk comes with static IP address and it's connected directly to the STC company. I need an expert who did this STC SIP trunk connection many times. I need him/her to send me the full configuration steps so I can do it. Note: I can not let you access the Elastix server so Please do not ask for that.
Hi arodrigue, I noticed your profile and would like to offer you my project. I want to make a phone verification system using asterisk or freeswitch. We can discuss any details over chat.
UI/UX Developer Web Developer Gamification Developer Asterisk Developer (VOIP, 911, PIDF-LO Stack, Dynamic Status) IOS Developer Droid Developer – NEED ASAP – current developer is leaving at the end of March Back-end Developer (need 2 of these)
Будь ласка, зареєструйтесь або увійдіть в систему для перегляду деталей.
Hi. I need a push service for asterisk to support Android/iOS softphones.
...Cuando intento hacer una llamada sale pero hay que intentar muchas veces, cuando sale la llamada si cuelga el agente la llamada del cliente sigue activa en el celular como si el agente todavía estuviera en línea y si al contrario cuelga el cliente en el goautodial sigue como llamada activa para el agente, es como si el sistema de colgado y despege del canal no funcionara. Uno de los mensajes de asterisk cuando se intenta hacer llamada y no sale es "everyone is busy/congested at this time". Quiero saber si me puedes ayudar a solucionar este tema....
I need someone who has multiple experience with all web programming and development. I have a lot of small task. So easy money for experience web developer. Skill Need: HTML, PHP, Laravel, Codignetor, Wordpress, Magento, Prestashop, etc. VOIP, Free PABX, ASTERISK, Linux, Centor, Ubontu Rest will discuss via chat. Write IDOUNDERSNAD then I can understand you read my project details. Note : My every task will be for 5-15 minuits for the one who has sufficient knowledge. AND payment per task will be 5-10. I have approx. 60 task. SO i you agree then bid other wise please dont bid. Thanks
We are looking for Asterisk developer to develop IP PBX and Dailer.
...to configure asterisk real-time to work with multiple tenants (e.g with tenant based contexts or dial plan aliases). Infrastructure exists in AWS for Database and auto scaling group of asterisk servers but the whole thing needs configuration. This is entirely from scratch. What we have ------------------------------------------- We have a fleet of raw, unconfigured asterisk servers in an auto-scaling group (ASG) in AWS. We also have a database instance in Amazon RDS - again, just the instance - no schema or configuration yet. We have all of the necessary networking infrastructure all configured in AWS with security groups, VPC etc... How it will work ------------------------------------------- Customers will have their own virtual PBX within this set up. We want ...
...access would be blocked if the client who wishes to contract the service does not confirm. * From the administration page you can go with a click to the client panel (without authentication) * pop-up window of incoming calls with contact details, if any, to the extension. There are already commercial solutions at this point, but we need adaptation to our project. This is integration with Asterisk, although you do not need to know Asterisk or the SIP protocol, since the program will receive the events of incoming calls with the necessary parameters, such as tenant and caller id. You can choose the event-oriented language you know. If you wish, we provide the server for development. We know that the solution is developed and commercialized, but without access to the...
Develop an inbound telephony script for asterisk What you will get - A machine with Asterisk installed, SIP information for the telephone connection, and MRCP servers installed for Speech to Text and Text to Speech. You will also get access to a web dialogueAPI and the input-output specifications of that API. What you need to do: Write a script that works according to the attached rules. Please read the attachment before bidding.
need to register 2 extensions as PJSIP on isable asterisk
Looking for a programmer that know how to create Amazon Alexa Skills as well as knows VOIP. We need to create skills to connect to our custom Asterisk PBX systems to support similar features in the Fortivoice skills app currently in Alexa. Please unless you have does this type of work before do not post. The more I can be hands off the more I am willing to pay per hour.
I have special dial plan, I need to fine tune these dial plans ,also need to extract required budget is 50$ fro this project ,please do not bid high amount I will not entertain such bids, if satisfied with work will give more work .
Require to install Asterisk + FreePBX on a tower server remotely and configure the SIP Trunk, Extensions, Agents.
Need to create a campaign manager as predictive dialer integrate with cloud & on-prem deployable with soft phone, we have a own contact centre platform as agent desktop solution where the calls should land on our platform . If any one have outbound dialer ready solution also fine.
We are looking for a consultant that is quite familiar with Kamailio and Asterisk. We are looking for someone that can build out an instance of Kamailio on one of our development servers which will be the public facing proxy. On the same server will be Asterisk running on a different port. We are hoping Kamilio is setup for best practices for security to prevent toll fraud, take advantage of any services that might have black lists or things of that nature. Have some security built in for blocking non-legitimate traffic, banning IP's or not responding.
...can our application push information to the desktop of agent who answered the call? Also, the information captured by the IVR would need to be sent to our application at the beginning as part of this session. Possible future milestones: ----------------------- (1) Help with a setup where we can plug into an existing contact center (such as Cisco, Genesys, etc.). It will probably need to use Asterisk to be added into a conference call at the call initiation between customer and agent. It will need to receive identifying information plus the media stream that will need to go through ASRs and the ASR output will be routed to our application as before. Our application will then need to update agent desktop used by that contact center. We will need guidance /consulting for architect...
Hi, We want to add sip phone to our CRM solution. Our CRM solution is web based and developed using by php, node.js, bootstrap. We are still working with some softphones (microsip, eyebam, xlite) by using Asterisk AMI. But we need sip based web phone and it must work with all voip software. We need strong documentation and easy installation. Our OS is centos. Thanks.
Hi, I have an Asterisk server running a small telecom operation. I reached a number of users that is beginning to affect Asterisk, so I need a more versatile SIP router. I would like to maintain asterisk to implement IVR logic, voicemail, conferences but I would like to pass functions like SIP contact management, authentication (Invite and Registration) and even subscriptions to either opensips or kamailio.
I need to install and deploy Asterisk projects on my server.
Knowledge of how to distinguish Asterisk call records and differences between queue records and call records and call transfers are required, please don't apply otherwise. Need a simple web-page based on PHP that we can have connect to the asterisk mysql database tables (remotely) on our FreePBX system and offer a date range (Default being the last week) to choose and a list of users next to their EXT #'s next to it and allow selecting multiple users . After submitting , a table would display with the following stats for each user selected... If it helps in anyway, we have Asternic Queue Stats Pro installed. Accuracy is paramount. # of Incoming Phone Calls # of Outgoing Phone Calls # of Phone Calls Missed Duration of Time Spent on Phone in min:sec # of Cal...
DLS Internet Services is a small Chicago-based VoIP service provider. We offer hosted PBX service that utilize our PHP-based software that is based on Asterisk 11. DLS is looking to augment it's team with a VoIP developer who can help moving our product forward and help integrate various 3rd party solutions into our platform.
Freelancer Project details The project is a videoconferencing system using PHP and jitsi. It will require development to use jitsi apps as native, or our own apps based on jitsi opensource code. We will leave the joice to the developer to advise. Background Zoom () is a full fledged video conferencing system built on top of jitsi video and offers meeting and chat, webinar, and conference room systems. Jitsi () is an opensource videoconferencing software designed to hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom
Freelancer Project details The project is a videoconferencing system using PHP and jitsi. It will require development to use jitsi apps as native, or our own apps based on jitsi opensource code. We will leave the joice to the developer to advise. Background Zoom () is a full fledged video conferencing system built on top of jitsi video and offers meeting and chat, webinar, and conference room systems. Jitsi () is an opensource videoconferencing software designed to hold a small footprint while providing meetings, conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom
Looking for a programmer that know how to create Amazon Alexa Skills as well as knows VOIP. We need to create skills to connect to our custom Asterisk PBX systems to support similar features in the Fortivoice skills app currently in Alexa. Please unless you have does this type of work before do not post. The more I can be hands off the more I am willing to pay per hour.
i have a call management panel where i can add client add prefix add gateway add dialplan then if all valid call will be passed now i need to filter calls like if a call come to our server u need to check its new or old if new then all call pass to a free dial plan if old then u need to pass the same dial plan its an call filter system i will give u a panel link where u can check everything before u start i have to know u have good Knowledge about voip and PHP do not place automatic bid only if u can make it then place bid thank you
We are looking for a SIP Load Balancer based on extension range. Ex. Extension 2000 -2499 goes to server A - Extension 2500 - 2999 goes to server B and so on. Server A and B are Asterisk Server. The SIP Load balancer must run in a Linux Centos Environment. Documentation on how to change Load Balancing Parameters is required.
Hello, I need someone who has good knowledge for FREE PBX. I need someone available right away and will keep for support in the futur. Who can fix the following problems : - Incoming number do Ring but then disconnect when pick up - Outgoing calls numbers show the General number and not the one encoded of the extension. - when you a forward to a Mobile Phone directly, it's the General number that is show, I would that it's caller ID that is shown to the person that the call has been forwarded. (Diversion Function). - Check that the best protocol is used for the phone. G929 ? - Security check that fail2ban , etc.. Regards
hi, i want to set my asterisk server to receive calls from another asterisk server which will then pass it to sip
good evening, im trying to use a proxy for my asterisk server to use the different ip addresses between my server and the provider terminating calls is that possible ?
good evening, im trying to use a proxy for my asterisk server to use the different ip addresses between my server and the provider terminating calls is that possible ?
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application 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 ...
