We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux
Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone
1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.
PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done
...with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined Deadlin...
Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets
...411 We use "9" to get signify an outside call. So when dialing a 10 digit local number we want 9XXXXXXXXXX sent to the PBX as it will strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a ...
...you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL] up to you as you like i just need pass a call on g729 codec under 15KBPS now as you
...A: On Cloud
And SIP Trunk has been created
SSL is configured by "Let's Encrypt"
Server B: On Local
And SIP Trunk has been configured
SSL is configured by "Self Signed"
Server A - SIP Trunk is appeared as not registered
Server B - SIP Trunk is appeared as registered!
I need to get all Registered, and the call is go through
...use Call Centre CRM. Get back to me with demo. Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Cal...
Following are the requirements: Good Communications Good Internet Speed Goal Oriented Mindset Team Following would be provided : CRM DATA Dialer VOIP Traning Support Team has to adhere to timings for reporting , Meetings, login time .
1. Call Durations: every minutes of calls will be count (outgoing calls only). total calls duration will be readable from web. 2. USSD Blocker: Any USSD or Balance updates USSD popup needs to be store and count first for calculation (its need to read balance update from web) then any USSD have to be removed from screen by App after 2 second.
International Call center campaign : Australian Solar appointment setting campaign. Per seen appointment $35 AUD. Sale close by client commission will be your. Centers just have to book appointments. Data will be provided by client. Training will be provided by client Directly in your call center. Dialer and voip provided. Long term campaign
Following are the requirements: Good Communications Good Internet Speed Goal Oriented Mindset Team Following would be provided : CRM DATA Dialer VOIP Traning Support Minimum Five agents are must to login . Team has to adhere to timings for reporting , Meetings, login time .
I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source
...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www
Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples
the project is to build ionic app plugin out of linphone sip framework or linphone-cordova-plugin. Plugin should have following features. Audio calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Call History Echo Cancellation Call quality indicator Secure user authentication : md5 / SHA256
We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.
I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with
I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else
...The application will feature voicemail detection function where an outbound call is placed from a csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should
we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion
Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)
...SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice
...have connected a VOIP DEVICE i want to send call THIS VOIP DEVICE now i have 2options 1) Install VPN on server and openwrt ROUTER and bridge network and set a static IP same serial on VOIP DEVICE useing MASQUERADE 2) make remote connections and USE 2nd IP on my voip device i need a solutions to pass call server to local VOIP device i cant use any PC
"HOME ALARM SECURITY OUTBOUND PROCESS WITH DDV" ****Only Indian centre**** **Only centre with past experience in usa voice proces...centre who can join the training schedule for the same process from next Monday 6:30 PM onwards inbox me. *No upfront or security deposit. *No any hidden charges. *Data and dialer provided. Ping me ASAP. Thank you.
...dongles. 9- set a call timout to avoid voicemail and its charges, also those timed out calls should be considered as not completed, not answered with 00:00 duration which affects acd ( i have this problem). 10-just click beside the dongle active call to spy on it(quality control) ( i have it done just need it in GUi) 11-set dongle call duration limit
...from [login to view URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested, he presses 1 and the call will transferred to an available agent. b- Vote/Feedback/Survey: a pre-recorded
...ask couple additional questions about the industry provided in the script. Average time per call 2-3 minutes. All software infrastructure (CRM+Dialer) and lists of contacts will be provided by the company. Technical requirements: Stable 10 Mbps or more Internet connection speed Windows/Mac computer with Google Chrome browser USB Headset with noise cancelation
Need a browser extension or appli...understand when the agent is available, on do not disturb or on a call. Need the availability to move a call from one agent to another agent. All call details will need to captured - Date & Time, Caller ID (Number) Start of Call and Finish, Agent ID Needs to be able to work across SIP and PBX providers using TAPI
...interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation from development; - log from customers and admins about changes; - each user have your color/logo (size from picture
Phase 1 Need to setup asteriskNow with ippbx (tata telecom). Setting up sip trunk. redirecting calls on specific numbers to specific sip no. At specific work hours or if specified through API, then to redirect calls to Mobile no. Setup recording of calls.
...using Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have