To set up an eCommerce website successfully web need to keep important point which can help to grow e-commerce business.
To invite everyone around Little Alden and Spring Lake Sunday December 11th, between 1pm and 4 pm At Cindy and Rod's, 3296 N Little Alden Lake Road, 218-391-5815 We'll have Fires going...we'll have Hot Chocolate to sip on and all the ingredients for you to make Smores ! Bring any Beverages that you'd like !! Sit around the Fires, get Cozy in the Bunk House or Relax in the Cabin Wear Some that is all Christmas...A Hat, a Sweater, some Socks...We'll try to get some good Pics to put together in a collage !! We hope to see you Sunday ! We just need a basic flyer to distribute around the lakes and to email. Use some Christmas colors...make it fun... I am looking for an inexpensive flyer that is professionally done..." not in my handwritng " Than...
...need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through WhatsApp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows WhatsApp executables, or by using the Android / Windows Phone mobile versions of the application, no matter the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project, we'll select the one, that triggers successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to the WhatsApp gateway 2) WhatsApp gatew...
requiero un manual o tutorial para implementar seguridad en el servidor asterisk, ya sea con fail2ban, iptables. para mitigar los regitros sip, ssh y los ataques de denegacion de servicios. Acepto cualquier sugerencia para mejorar dicha seguridad. I require a manual or tutorial to implement security in the asterisk server, either with fail2ban, iptables. to reduce sip, ssh logs and denial of service attacks. I accept any suggestion to improve said security
1. Vendor wi...of Virtual CUCM (Unified Call Manager) on the Virtual Machine, The VM will be provided by the Customer on their ESXi license 5. Perform Standard & Advanced Configuration of Cisco UCM 6. Implementation of 30 qty. IP Phone 7. Rack Stack, & Power of New Voice Gateway Routers ISR4331-V/K9 (Qty. 1) 8. Initialize Voice Gateway Router • Perform staging & Software Upgrades • Configure OOB management 9. SIP trunk established between Cisco UCM & Voice Gateway Router 10. T1/E1 termination and its configuration for outbound call 11. Configuration of logical partitioning in CUCM 12. A-Flex-3 Smart license allocation 13. Overall IP Telephony system testing & go-live of the deployed solution with Customer 14. Submission of Implementation document ...
MicroSip - is an open source platform that can connect to SIP central. - can do audio and videocalls. - can create buttons (shortcuts) to start a videocall. fig - audio calls can be started from html. (command line) - can't start start video calls by command line. (NEEDED) I want to create a php page with buttons that can start a videocall by MicroSip (or others) fig. butler On the desktop (device with te buttons) the videoscreen has to be hidden when starting the videocall.
We installed goautodial v4.0 from iso Kamailio running HTTPD OK SSL certificate OK RTPENGINE Ok Our main issue is the following: 1) Agent need to press (Login to dialer 2 times) 2) Can't register GoIP gateway (SIP Trunk) 3) Can't hear any voice. Only Goautodial V4.0 specialist is required...!!
Im looking for someone to build be a self hosted voip system We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc'
Hi, We need someone to design a brochure for us. The brochure will be sent to companies to introduce our paint and sip events. Most of the photos will be provided + a sample. However, we need a creative person who can design something unique, very clean design and professional look. Photos will be provided, the person might need to write some parts by him/herself. We need someone who have a background in marketing.
webRTC SIP / Chat / video / conferencing / Electron / typescript Android / IOS app need to develop a app to work in electron / android /ios / angularjs
I'm looking for someone to develop me an Android app that acts as a ""SIP server / mobile gateway" and makes the cellular line of the smartphone usable with a SIP client. So it is also a kind of VOIP mobile gateway. It may be possible to port the free IP telephony software to use the cell phone as a cellular gateway. However, it is important that the APP serves both as a gateway and as a SIP server, so you can log on to the app with any VOIP client to make calls via the smartphone's cellular network using SIP protocol (IP telephony). Important: The app should run on any modern Android smartphone running Android 10 or later.
I would like to have an artist that can paint up a picture in 30-45 mins. I would like this to be a surprise proposal where the painter paints a picture of me kneeling asking her to marry me while pretending for us to hang out and sip wine. When the painting is revealed to her she will be surprised and turn around to see me kneeling. I would ideally like this artist to be able to do this at a winery in Temecula.
Hello We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc' The end goal is to be able to start selling voip services. We prefer a person who did something like that before. I will be happy to answer questions.
Make PWA SIP-client be able to receive calls at lock screen mode ( iOS/Android smartphones). I know the current problems with push on both platforms :-) But Im looking for solve it. And want to pay for solve.
in voip for USA/Canada LRN/LNP query is conducted to get the shortest path and query server send SIP 302 redirect messages as required. We will update its database manually but it has to respond to query's fast and accurately as per pre-loaded data. Only knowledgeable person in voip architecture should bid to this project and i will pay only bidding price, no negotiation on price shall be done after bidding.
Hi, I need someone who has an excellent experience in the FREE Open-Source SIP SBCs to suggest the vendor we will install and help me to install it Kindly don't bid if you don't have the experience that is needed, Thanks!
Necesito una persona con conocimientos de python, android, voip que me desarrolle un modulo de sistema que me que me valide mediante una llamada a los usuarios de mi sistema, cuando mi sistema le envie un codigo sms, lo validen marcandolo en la llamada, esos numeros que marquen mediante la llamada, los guarde en el sistema... el sistema tiene que tener TTS (text-to-speech) para que recono...desarrolle un modulo de sistema que me que me valide mediante una llamada a los usuarios de mi sistema, cuando mi sistema le envie un codigo sms, lo validen marcandolo en la llamada, esos numeros que marquen mediante la llamada, los guarde en el sistema... el sistema tiene que tener TTS (text-to-speech) para que reconozca la señal de los numero, a parte que permirta hacer conexión VOIP con...
I have Windows 2022 running DHCP if you need to configure option 150 and we can install TFTP server to upload SIP firmware
i have accounts with twilio and signalwire create me a a sip trunk configured to work on my server only outbound dialing for gateway (termination only) must see calls going out live on my server to get paid this should be easy money for the right person need this for now, contact me for additional details ( must know astpp and goautodial v3 ) *******dont ask for other payment method dont ask to talk on different platform dont ask to avoid freelancer fees!!!!!!!!******
Hi, i am looking for someone that can assist me with some small setups on vicidial. I have already setup a vicibox solution on the server and have a sip trunk. We did use fusionpbx for a small call center with only outbound calls but looking to move to vicidial. I am struggling to setup the carrier and have issues with the time sync on the vicidial agent side. Our time is out with 10min for some odd reason. I have already checked the files in apache and cli which is correct. the settings in vicidial is also set to +2 for time zone. I am looking for someone that can assist just to setup the sip trunk and also make sure that the agents can dial out with the webphone that is built into vicidial for now. Will you be able to assist today and what will the cost be.
Customer installation consists of a 3CX PBX, Patton T1 VoIP gateway with 2 T1 connections for a total of 46 channels. Customer is experiencing that some incoming calls drop after a few seconds and 3CX reports "route busy". Have sent Wireshark captures to 3CX and Patton tech support and unable to resolve the issue. The problem could be the T1 provide...installation consists of a 3CX PBX, Patton T1 VoIP gateway with 2 T1 connections for a total of 46 channels. Customer is experiencing that some incoming calls drop after a few seconds and 3CX reports "route busy". Have sent Wireshark captures to 3CX and Patton tech support and unable to resolve the issue. The problem could be the T1 provider. Need a network engineer to work remotely capturing SIP data and deter...
WHAT YOU WILL DO: · Will work on complex assignments and perform a full range of technical support activities. · Acts as a member of quick-response support teams to meet customer deadlines · Operates independently to resolve complex design/configuration problems. · Has a wide degree of creativity and latitude in SIP/RTP technology. · Has ability to plan, implement or oversee coordinated testing of new applications, to ensure compatibility with existing hardware and software applications. · Responsible for engineering and/or analytical tasks and activities associated with areas within the telecommunications function (e.g. engineering, implementation, diagnostics or operations/user support). · Monitors the operation of telecom (P...
...Technical Skills Relevant experience in Android app development Proficiency in Kotlin (must) and Java languages to write clear, readable, and maintainable code Experience in Android components Knowledge in Jetpack components Hands-on experience in mobile app architecture, design patterns and fundamentals Familiar with RESTful APIs and XML to connect the app with backend services Knowledge on SIP, TLS, IMAP protocols preferred Experience in Crash Analysis, Push Notifications (FCM) Ability to work as a group contributor or independently if needed Hands-on experience with Android's debugging, unit-testing, memory and performance optimization tools. Problem-solving mindset, analytical abilities, strong technical and communication skills Other skills 5+ years of expe...
Hi Farrukh O., In your SIP Phone project we need you to change some codes for debian (armhf). In current project all is working. But microphone and speaker are not working. We need your support on this issue. Thanks.
Looking for a server admin with experience in Windows Hyper-v / TrueNAS / SIP / SBC / PBX to help with management and setup for our server cluster as well as voice customers. Skills in networking including routing, BGP, VPLS, Mikrotik etc would be an added bonus. This would be a full time position covering overnight USA time zone.
Looking for a server admin with experience in Windows Hyper-v / TrueNAS / SIP / SBC / PBX to help with management and setup for our server cluster as well as voice customers.
I need to create an Intercom App for a building. There will be: - A searchable List of Residents - A camera that will scan QR Codes - Sip Softphone that will call the residents. We will need to handle the key response DTMF to open the door if the resident presses on *0
Hello, We have a VoIP SIP Soft-Phone that uses WebRTC & JsSIP, our dial pad is not working correctly to enter extension #'s for the DTMF tones, also the audio quality is not the best and we need to fix that too, in the past it was a lot better, not sure why the change, the hard phones do not have these issues, just the soft-phones. We need an expert who can solve these issues, please do not waste our time if you are not an WebRTC & JsSIP expert with VoIP / SIP phones. Thank you!
I am looking for someone to make a simple call flow in bound dial plan for asterisk, should be as follows . call to did - dial plan will send the call to whichever agent that is free and logged in sip account
Avaya Sip Trunk Configuration (Inbound/Outbound Route)
i would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in to it use a SIP software in order to answer customer calls. What I want is a convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field and last three data pulled out to select to go into the dispatch panel Deploy PBX Create few extensions Create script for take CID of inbound calls check last 3 records, sent CID and last 3 records via api to dispatch app Help to setup tranks and dialplan
Develop a sip Voip test application: a command line client in golang to connect, receive and place voice calls using sip services like sipgate or linphone. Must run on Debian . Audio interface must be microphone, speaker and file. A local file will be played to the destination and will be recorded at the other end. Expected minimum functionality: Register messages Place call Pick up call Play audio file Record to file Dtmf Select audio codecs Log all events, timings and network information (start/end of the call, data rates, packet information, loss, jitter, used codec …)
i would like a asterisk and sip software install on my laravel webapp so Each operator is loged into it with an extension is assigned an extension and loged in to it use a SIP software in order to answer customer calls. What I want is a convenient API to interact with asterisk server, for example when the operator receive a customer call, the caller id being inserted in a text field amnd last three data pulled out to select to go into the dispatch panel Deploy FreePBX Create few extensions Create script for take CID of inbound calls check last 3 records, sent CID and last 3 records via api to dispatch app Help to setup tranks and dialplan
Hello , We are looking for the expert who can develop the Sip Mobile dialer for Android and Iphone Application and Sip Voip Tunnel , Where sip mobile dialer and sip voip tunnel to be integrated and to work in gulf countries as well
I have just launched a website for several calculators - Percentages Calculator, EMI Calculator, and SIP Calculator. I am looking to get some content done on three pages with at-least 1500+ words on each page. I will give you the link to my website and the competitor pages once I finalise you. If needed, I can also give you the keywords.
Scope and goal of requested alterations is to have a complete set of prints ready to submit to Grand Traverse County Building dept. 1. Add elevations that are missing 2. Add foundation drawing Indicate all point load and load bearing walls. 3. Change from SIP roof to conventional trusses for all but the great room and kitchen 4. Details of point loads and connections where required 5. Soil bearing capacity must be on drawings for sandy soils 6. Show the required connections from timber trusses and SIP panels to foundation or walls 7. Verify attached local design criteria has been met 8. Braced wall lines and design
Se requiere conectarse a una VPN para poder conectarse a un servidor dentro de la red con FREEpbx y configurar un trocal SIP con SBC con los datos que se brindaran. Crear una ruta entrante y saliente.