Freeswitch voicexml asteriskemplois
We have a following problem : * we are using AMI originate call to Q to dial numbers from a list (csv) * call is initiated from Q *when customer picks up call is connected via Q to first available agent we noticed that over 30 % of calls have status ChannelStateDesc : 0 (down) and we need to debug what could be the isssue. Although agents are available and in a Q, they dont get call always. im attching a pic of a channel state and we want to know what that means (is it a failure) since on a pic we see duration 11 sec and channel state down we are using Yeastar S300 PBX
Current Situation We currently take CDR’s from our voip platforms, freesbc and Xorcom (asterisk)and suppliers, our billing platform can only take one cdr input and can’t produce consolidated cdr’s for wholesalers or importing into our billing. What we need Design and develop a system that will have the following parts. 1. An importer – which will allow us to import CDR's and process the required fields that we want based on the different format types, it should then import them into our database in a common format 2. Mapper – We need to be able to map phone numbers and services to particular wholesalers and apply rates against those files based on a service type 3. Reporter – We need to be able to run reports to give both detailed and g...
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY.
I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination ...whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY. VOIP opensips CGrates, webrtc as...
Deseamos un desarrollador para el modulo de reporte en la plataforma vicidial, basada en asterisk para los que conocen, basicamente es modelarlo a una interfaz mas amigable luego realizar un click to dial desde nuestro sitio web para que el visitante lanze envie una llamada por esta interfaz. Ademas tambien incluir un boot que pueda interactuar con los db y pueda a traves de solo el codigo leer el registro y brindar informacion de cobranza (no lo queremos como audio sino que lo haga el boot lo realize). al interesado escribirnos por este medio
Cine este interesat sa instaleze, configureze si asigure mentenanta pentru o centrala voip Asterisk la care utilizarea o sa fie minima (1 sau, maxim, 3 trunk - uri + 2 - maxim 5 extensii), este invitat sa ne trimita o oferta. Pentru ca avem in vedere activarea callback, IVR, XMPP, etc. vom aprecia experienta dovedita pe astlel de proiecte. Multumim!
Hi Eremin P., I noticed your profile and would like to offer you my project. We can discuss any details over chat. we need a WhatsApp gateway via Asterisk, is this something you can do?
Hi Mario Felipe R., I noticed your profile and would like to offer you my project. We can discuss any details over chat. we are looking for an Asterisk to WhatsApp gateway, is this something you can do?
We have running FreePBX + Asterisk. We have tried to upgrade FreePBX and something went wrong and now our system is down. Can anyone knows about CentOs 6 32 bit version. Asterisk installation please reach us. Its very urgent.
Hi Sergey S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. What I want you to do? Create PBX server with SRTP, ZRTP and voice pitcher: I will rent OVH server, so thats not a problem. But I want to have fully encrypted VoIP Calls. Tell me price, and time.
Connect an outbound node ESL server with FreeSWITCH running inside a docker container. Also, configure fail2ban with IPtables for VoIP security.
I have an asterisk server currently set up which processes calls. I want to add the lines of code to enable the caller to play ON HOLD music by pressing a number which will play the music for the person receiving the call.
Asterisk based switching to mobile applications. We need a SIP to WhatsApp gateway. The Gateway should be able to pass voice calls incoming to the Asterisk server and terminate through WhatsApp to the called party. The WhatsApp gateway should return the proper signalling, Successful call, Busy, unavailable, if the called number is not available then the WhatsApp gateway sends back a 503 redirect.
We have to develop a VOIP application from LINUX distributed units and WINDOWS workstations. The LINUX units are based on a quadcore CPU . They are installed on vehicles and vehicles and are connected to a center through a 4G router. The operator on the vehicle has to use the microphone input and the audio outputs in order to speak with operators in a center. The audio output is on speakers , so it is a hands free application. In the center the operators work on a windows computer. We have to set up this system: programs to load on the LINUX unit and programs to load on WINDOWS computers. In the We have to run a VOIP application : the operator
Hello, i need to integrate Odoo V13 with Grandstream UCM 6500, the odoo crm, contact, leads, customers and more, should have click 2 call function, popup for incoming and outgoing call, new incoming calls to generating leads and more, call recording, call transfer, auto search phone number within odoo database for incoming ca...with Grandstream UCM 6500, the odoo crm, contact, leads, customers and more, should have click 2 call function, popup for incoming and outgoing call, new incoming calls to generating leads and more, call recording, call transfer, auto search phone number within odoo database for incoming call and auto opening the lead/customer/opportunity. If you know odoo 13 and you already develop/integrate asterisk phone system, please let me know. I need this job to be ...
Integration between zendesk and asterisk
- Mensagem de Bem-Vindo - Mensagem da Administração - Salas Gerais - Salas Privadas - Salas Numeradas - Sala da Monitoria - Pin - Torpedo - Lista de Participantes On-Line - Murais de Mensagens - Caixa Postal - Menu de Opções Básicas - Menu de Torpedos - Menu de Opções Avançadas - Menu Opções de Pin e Caixa Postal - Menu Trocar Senhas ou Gravação do Nickname do Pin - Menu de Caixa Postal - Menu Lista de Participantes On-Line - Recurso Salas Numeradas - Recurso Sala da Monitoria - Menu de Murais - Menu de Opções do Mural - Recurso Regras do Serviço - Enviar Uma Mensagem para a Administração - Cadastramento de monitores Obrigado! Nosso serviço ...
Tener experiencia necesaria para crear servidores de desarrollo y Producción en AWS. Experiencia en desarrollo de sistemas administrativos (CRM, Créditos, Tarificadores, Etc.). Experiencia en desarrollo y mantenimiento de sistemas con Asterisk, FreePBX, Issabel. Dominar el idioma español.
I need a shell script which I will call script user dst cid_prefix audio When I call the script like script user 123456 54 audio1 ip it means Dial 123456 your cid is a random 10 digit number which starts with 54 via SIP/ip play audio1 record entire call in a folder ./user also possible to call like --db user 10 [dst_prefix ] ip This will query db like select dst,cid,audio from db_table where dst like dst_prefix% limit 10 order random then dial the 10 numbers all at once
Most critical information has an asterisk; ideally, I'd like it all. This is for a national scrape -- listing in the United States on a website that has international reach. * Homeowner Name * Property manager * Address, City, State, Zip * Phone Number Email Address Bedrooms Bathrooms Square footage Home type? Swimming Pool? Cost: avg/night
Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. My budget is below $50 Thank you.
Buna, caut specialist pentru instalarea si configurarea unui server asterisk/freepbx ca centrala telefonica si cu dongle usb cu functie SMS-gateway. cu stima,
Hi We need someone who has knowledge in SIP , Asterisk to help us set up and outbound system. Please apply ONLY if you have relevant experience. Regards
Necessito de Operator Panel e Call Center parecido com 3CX para FREEPBX ISSABEL ASTERISK Necessito do código fonte.
Necessito de Operator Panel para FREEPBX ISSABEL ASTERISK Necessito do código fonte.
To make interchangeable Invoice and Credit Templates. The Templates are made by us with LibreOffice and must be changed to pdf without loosing calculations. They will be filled out on our server, to be send by email. *** Requireme...LibreOffice and must be changed to pdf without loosing calculations. They will be filled out on our server, to be send by email. *** Requirements: 1) LO Calc spreadsheets for addresses, with interchangeable background image of Invoices and Credit Notes. 2) Software to change the ods files to pdf, without loosing calculations ! 3) Asterisk CEL (not CRM) to be used, to fill Cells on spreadsheet. *** This is not a standard project ! Very good knowledge of Billing software and Asterisk PBX is needed. Good communications are necessary ! See attachme...
Hello / Good evening everyone, I would like you to help me create an asterisk call file which places several dtmf commands in the call file channel once the call is connected. The DTMF does not necessarily have to go through the call file. If you want you can use AGI, AMI and the file, but my goal is to place dtmf commands in an ongoing call generated by a call file.
Full Asterisk IVR Setup training and cdr report with below requirement 1. Creation of IVR flow chart and diagram 2. Communicating with other stakeholders for the Audio files. 3. IVR Configuration on FreePBX 4. Setting up Inbound routes to point to IVR (via GUI and Command line) 5. Other tweaks on IVR (schedules / time conditions) (via GUI and Command line) 6. Setup of dial plan on GUI and CLI 7. Setup of IPtables and Fail2ban on CLI 8. Install and configure a complete fully functional PBX 9. Troubleshooting 10. CDR Reports formating 11. SIP Configuration 12. Asterisk API 13. Extensions configuration 14. Queue Configurations 15. Webrtc 16..Robo call using CLI 17. Single auto call using CLI 18. FreePBX SDK
We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of telephony systems based on these platforms -Excellent understanding of pbx telephony, dial-plans, security -Excellent understanding of hosting technologies -Excellent understanding of Database (SQL) with High Availablity (HA) & data replication/sharing, DRDB IF YOU HAVE NOT DESIGNED AND DEVELOPED A SIMILAR SOLUTION, do not apply. we require an experienced team capable of full delivery on time and within budget. The initial scope of project will include creating the following op...
Hi, We are using asterisk server and g729 codec. It supports only Pass-Threw mode because it's the free version. But we need a free version of this codec that supports Transcoding mode. If anyone have that please knock me. Thank you.
To make many Invoice Templates for inbound DID billing. Open source software is to be used for this project. The Templates in pdf are to be filled out on our server, to make the invoices to be send by email. Required: 1) LO Calc spreadsheet with interchangeable background image...many Invoice Templates for inbound DID billing. Open source software is to be used for this project. The Templates in pdf are to be filled out on our server, to make the invoices to be send by email. Required: 1) LO Calc spreadsheet with interchangeable background image of Invoices. 2) Software to change ods to pdf without loosing calculations. 3) Asterisk CEL to be used to fill Cells on spreadsheet. Very good knowledge of Billing and Asterisk PBX is needed. Good communications ! See attachme...
Hola como estás? Quiero instalar chan_dongle a mi sistema asterisk para poner en funcionamiento un moden huawei E303. Podes ayudarme?
I need updated chan_dongle for asterisk 16 or newer
We have ASTTP voip billing software and Free switch running on Asterisk. We have many hack attempts everyday when we open port 5060. If we keep the ports 5060 and only open other ports such as 2070, then the hack attempts are down to 1%. We want to keep port 5060 open, so we need someone to help us, how to keep safe from the hackers on port 5060. Also give training for future security treats
I am having problems with extension loosing register, i dont have a very good internet connection but i am almost certian that i can fix this problem, customer has 50mbps upload and download, the problem is the pings not being very stable, due to a wireless ptp connection, freepbx / asterisk setup.
I need a logo...temperature of people walking in and to detect if they are wearing a face mask through the use of AI. Can you provide me with 4 different concepts for the logo. The colors we are leaning towards are Blue, red, green, white. Some combination of those. Looking forward to some creative ideas. I've also attached our corporate logo one of the ideas i would like to flush out is: Health (asterisk) Guard For the asterisk i want something like this: but instead of having a snake in the middle we should have something like this: Logo needs to be vector. Please provide all source files and fonts.
Hello, Looking to get a previously working script modified/updated to work with new underling change. The script was written to work with Asterisk 11 and we are not running asterisk 13 and due a change in code the script no longer works. Phones are SPA5xx The scripts checks for calls that are parted in a specific parking lot and then displays all the related parked calls. the script is attached The old asterisk command to show parked calls was parkedcalls show The new asterisk command to show parked calls is parking show output for new is: Parking Lot: parkinglot_7 -------------------------------------------------------------------------- Parking Extension : 1170 Parking Context : parkinglot_7 Parking Spaces : 1171-1180 Parking Time : 24...
Need someone available in KSA Riyadh, We need to configure the Yeaster Part .
a) on the menu for the new registration of a listing, i want the button "show all" to be activated from the beginning b) on the search menu (under the rent/for sale bar), i want a new bar to exist having the Home/Store/Office/Land/Rest filters. c) the status field (on new listing menu) to be mandatory. Also i want all the fields which have an asterisk to colour the asterisks with red colour d) the other contact selection (on new listing menu) to be acitivated from the beginning (the field other contact phone to be mandatory as well).
Мы, контакт центр. Используем в данный момент СРМ известной компании, но хотим свою. Есть не большой сервер на котором установлена Issabel PBX, требования - настроить её работу полностью. Подключение сип транков, маршрутизации, очереди, звонки внутри и за пределами сети, вообщем все! В дальнейшем нам нужен будет человек который будет её обслуживать, если вы можете это тоже делать, то пишите, обсудим на каких условиях и нюансы работы.
hello i am looking for someone with freeswitch knowledge person with astpp (billing) will be very good. i need something to do with SIP TLS with letsencrypt and other voip work.. so you need be very good on freeswitch for this
Hacer modificaciones a la plataforma de Issabel: 1.- Poder identificar quien corta la llamada, cliente o agente 2.- Marcar hasta 4 números por registro 3.- Mensaje emergente cuando el agente recibe una llamada de una cola especifica 4.- Hasta que el agente guarde datos le caiga la siguiente llamada 5.- Bajar el tiempo de revisión de llamadas agendadas 6.- Poder tener hasta 3 llamadas activas (entrada, salida y agendada) 7.- En las campañas de salida, grabar en el archivo csv la fecha y hora de todos los status 8.- Agregar sonido cuando el cliente cuelgue
Looking for Asterisk expert who can help configure PBX. This task probably needs LESS than an hour work in this case my budget also mirroring. You need to able to talk.
We've created a platform where a presenter live streams video and voice over WebRTC from his mobile ovr cellular network in the field to audience at their homes that can only use voice to ask questions. The whole process works on browser only and is global (both the presenter and and the audience can be f...his mobile ovr cellular network in the field to audience at their homes that can only use voice to ask questions. The whole process works on browser only and is global (both the presenter and and the audience can be from anywhere in the globe). We need help with twicking our solution to get the best video quality we can, giving the circumstances. At the moment we are using Kurento, red5 and FreeSwitch (BBB). We are looking for a real WebRTC professional with vast knowledge ...
as title need asterisk guru support for a call center project and little small distribution
We need to be able to 1. allow the user to call the lead phone number, call the contact in opportunities, call the account 2. call a campaign Please see detailed explanation. (sorry for double posting - I could not change the file in the other project so I had to close it. sorry going nuts on this. Trying to clean up and all of a sudden I cannot talk to people)
hello i want some one to write java script code to make web rtc call so i want button in my site to make sipjs call to asterisk
We need to be able to 1. allow the user to call the lead phone number, call the contact in opportunities, call the account 2. call a campaign Please see detailed explanation. (sorry for double posting - I could not change the file in the other project so I had to close it.
We need to be able to 1. allow the user to call the lead phone number, call the contact in opportunities, call the account 2. call a campaign Please see detailed explanation.
