Asterisk cisco sbc sip rtp işler
i have a simple asterisk set up where users can connect with sip phone and make a call, i want to temporarily change the dial plan so that users can just hear a audio that i saved in sounds when they try and make a call
Simple configuration project for Extendeded services system.
We are building a simple business communication app for entrepreneurs. It is something similar to Google Voice. We have built most of the features except certain aspects of the call features itself. Who we are looking for: 1. You have a general understanding of voice communications (SIP call flows, WebRTC, SIP over WebSockets, and so on) 2. You have more than 6 years mobile app development experience. 3. You have more than 2 years experience with Flutter. 4. You are comfortable with the GetX package for routing and state management. 5. You are able to write platform-specific code (Kotlin and Swift) if necessary. 6. You speak English fluently. What we expect 1. Fix bugs and respond to support issues
We need someone to service our Cisco phone system. Can be done remotely. Overview of work needed: -Change main menu -Delete unused mailboxes -Reset voicemail boxes -Restore access to control panel -Tidying up
Hi Abu K., I need some help to configure a phone survey on a client's Asterisk server and integrate the results with Stats () Can you help me with that?
Need to swipe this switch - Upgrade this switch - Configure 1 VLAN - Make all ports active to the VLAN - Configure fibre ports on that same VLAN IP Pool make accessible via WEB IP address Host IP address I will be onsite today if you need any thing done this has to be done today around 1 hours Work
• Logo design for contracting firm specialized in Electric contracting. • Company Name , Sheikh El-Balad شيخ البلد SBC • Logo should be creative, new , catchy • Color mood ( Light Blue, Orange, Grey) one or two colors max • Logo must present only in white background with NO effects • Logo should be elegant and modern
Cisco FirePower to ISE integration with AD/AUTH realms for VPN & Wireless Policies
Hi I need you to design a storyline for a promotional video of my event. the event is international accounting youth camp that will be run for 5 days. some information about this event are: - target : 100 undergraduate accounting students all around the world - programs included: 2 webinars, 2 academic-based event, 2 fun-based event and a debate competition - platform : virtually through cisco webex meeting i need an attractive storyline that can attract people to watch the video and join my event duration of video is around 2-4 minutes.
I want a functioning network to work with BGP, OSPF and EIGRP. Everything is given in the doc below. Working with Cisco CCNP. All the commands used should be provied to me in a doc file. IPV4 and IPV6 Addressing should be provided along with implementation document.
...understanding of the version control Git tool. Understanding and Strong skills in server-side programming skills - concurrency, network, socket, memory. Good knowledge of SIP. RTP protocols and stacks. Knowledge of development in Linux. Solid understanding of the SDLC Participation in open-source project layouts. Reviewing, testing and debugging codes Constructing database architecture as required Collaborating with interdepartmental teams for product release schedule and ensuring timely deliveries. Schedule product releases with internal teams. Required Skills Demonstrable experience and understanding of Erlang, OTP, SIP. Understanding of Event Loop architecture. Experience with asynchronous programming in JavaScript Some knowledge of basic frameworks such as HTML/...
THE INSTRUCTION OF THE SCENARIO IS LABLED IN THE .pkt FILE. YOU SHOULD SUBMIT ONLY THE CONFIGURED SCENARIO IN THE SAME .pkt FILE THAT ATTACHED HERE.
Installation and maintenance of an Asterisk telephone system. Takeover of the old Asterisk system. (The old system is 15 years old). Many dial plans need to be rewritten. New functions are to be added, e.g. chat rooms for conferences etc. Provision of the data e.g. on an ASPX interface for other systems. We would like a contact in German. --- Installation und Pflege eines Asterisk Telefonsystemtems. Übernahme des alten Asterisk Systems. (Das alte System ist 15 Jahre alt). Es müssen viele Wählpläne neu geschrieben werden. Es sollen neuen Funktionen hinzukommen z.B. Chaträume für Konfenzenzen etc.. Neue SIP Telefone installieren. Intergration eines SIP Anbieters wie SipGate sowie weitere Verwendung des ISDN Adapter. ...
Hi Umair Y., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
We need an expert on Cisco ACI platform who has real-word experience in design, implementation and operation of Cisco ACI solutions.
i need to detect a loop mac address in cisco because my provider told me you have a loop with this mac xxxxx any help
I need someone who can sucessfully setup Asterisk and Google Speech to Text. More addons will be done later which is payable extra.
Install TLS 'let's encrypt' certificate, setup PBX to use TLS with SRTP
1. Conducting Technical Interviews for following L2/L3 Profile 2. Recording the Online Interview Session 3. Sharing the Interview feedback in a prescribed format Profiles for which Interviews are to be conducted 1. Windows 2012 R2 or Higher , DHCP /DNS etc concepts , VmWare ESxi / VCenter - L2 / L3 2. Network and Network Security - Cisco and Juniper - L2/L3 Profiles 3. AWS / Azure System Administrator - L2 / L3
Hi Harold B., I have some problems with my magnusbilling installation, especially SIP code 403 chanunavail. Can you help me to solve it ?
Need someone with good skills in Computer networks and Cisco packet tracer.
Need a softphone application with ability to scan qr code for login Customise caller id Hold music button And few extra stuff
We allready setup fusionpbx system with TATA sip line... We just want to configure call flow like all call equally divided to tha agent. And most important call forwarding on mobile and Don't call the agent who is busy.
Notre projet consiste à configurer un site-to-site VPN entre nos instance qui sont dans un VPC sur AWS avec un Routeur CISCO ASA5500 Series. Nous voulons le faire le plus rapidement possible et faire tous les tests de connexion à savoir : PING, Telnet
The project involves creating an outbound dialer software with IVR functionality.: • Develop a system that is capable of making bulk calls to recipients. • The system must be able to be hosted online and be able to connect to operator E1/SIP Trunk. • The system must be able to schedule phone calls. The system must have a phone book, that includes name, number and sex (male or female) fields, the most important field is the number. The ability to create a group should also be available. • The system should be able to create different Campaigns (Campaign creation). • The system should be able to create IVR Menu’s for specific campaigns. Example Campaign: Training people about financial literacy in rural areas, the course has 6 lessons which will be de...
Cisco packet tracer perform home automation of internet of things
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Hello Friends, I am looking for a full time VoIP telephony Developer who can create a web based SAAS Application for Call Management Portal that can help business to manage their huge call volumes. For more details, I can also show you the reference applications.
We are looking for Cisco ACI expert engineer, who could assist in a ongoing project .
Hi Arpit M., I have a MAgnusBilling softswitch based on asterisk and I have some issues that I want to fix. Can you help me for this ?
I have an in-house build custom CRM and Asterisk for call center. When there is an incoming call, I want a pop up on on agent screen showing customer info from our CRM.
I have installed magnusbilling which come with Asterisk 13 on Debian Need someone to configure Webrtc clients to connect to it useing web phones And provide a " how to " guide file Thanks
Hello , I am looking for someone who can install me a goautodial 4 with web rtc secure asterisk database and web
Hello Friends, I am looking for a full time VoIP telephony Developer who can create a web based SAAS Application for Call Management Portal that can help business to manage their huge call volumes. For more details, I can also show you the reference applications.
I have a small asterik module working on asterisk 13, I need to port it to asterisk 18
I'm looking for experienced web developer with WebRTC experience building dialing systems from scratch. We have a cluster of multi-tenant PBXs (VitalPBX) the new dialer must be integrated with. We expect this dialer to allow agents to place/receive calls, hold calls, transfer calls and any other normal call centre operation features to be implemented. We are open for the technologies you would like to use, e.g. it can easily be React.js with JSSIP library, or Vue with Sipml5 etc. It's would be also nice to stay in touch with developer for further development and some support in case we need something to know or to get be able to consultate with developer per hour/project basis. Minimum experience required: 2+ years web dev expertise with solid Javascript knowledge, WebRTC an...
Need to do network design in Cisco Packet Tracer according to given proposal
We are looking for an expert who can help us to make our webrtc client working with opensips. We have Opensips as SBC and FreeSWITCH to handle media + call routing logic. If we connect our webrtc client with FreeSWITCH directly then webrtc working well but when we connect the webrtc client with opensips then outbound and inbound calls are not working. Please bid only if you have worked on similar task.
Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards
Dear everyone I have my own FreePBX and Website. The Website built with Laravel. I gonna integrate FreePBX with Laravel. About Auto creating Sip Account, IVR, OTP Feature, I want integrate. Firstly, I want autocreating feature. so when user register our site, then sip account for the user will auto create in freepbx. I think we have to do this with GraphQL API or Rest API. if someone have experience in freepbx and graphQL and PHP, please send bid. This is long term project. Best Regards
Ayuda con algunas preguntas en tema de IT essentials cisco
Need a developer who can build me asterisk with ready chan_mobile module in puppy linux or ubuntu or any other GUI linux
Dear Everyone I installed FreePBX. and I have my website. I gonna integrate freepbx with my laravel website. To do this, I need professional freepbx expert. the expert should be get freepbx, asterisk, php, mysql skills. so this will be long term project. Best Regards
Buenas tardes, en estos momentos tenemos un Webphone desarrollo con la tecnologia SIPML5. El webphone funciona de manera correcta y esta desarrollado en javascript con .net. El inconveniente se da que a veces se suspende la sesion del explorador y necesitamos evitar eso. Entiendo que necesitamos un experto en javascript que nos ayude a ver este punto y que haya desarrollado un webphone antes.
Hello, we have a local running Vtiger 7.4 installation and also Asterisk with Freepbx. Over Asterisk there are 2 Snoms Phones connected. Vtiger Asterisk Connector is also installed. We need support to link it up, so that we can make and recieved calls on our Asterisk over Vtiger. Best regards
We are changing from one SIP provider to another one, and need configuration update. All the configuration is in place, and just need to change to the new one.
We are trying to apply the rip & OSPF protocols over the project network, but we encountered errors that we could not solve in the tables and transmissions.
We are trying to apply the rip & OSPF protocols over the project network, but we encountered errors that we could not solve in the tables and transmissions.
