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    8,499 simple sip program java jobs found, pricing in NZD

    We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

    $1093 (Avg Bid)
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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

    $222 (Avg Bid)
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    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [login to view URL] if you can make it please bid

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    Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want t...freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

    $337 (Avg Bid)
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    I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

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    ...create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want to make it under 10KB we need to remove or change RTP payload or you can make a SIP client to rerouting call change PTIME or if you want you can do anything else as you want we have only a openwrt router in bangladesh and you can install small package like any SIP CLIENT

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    I have a bunch of phones ...firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they come up and provision.

    $15 - $44
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    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

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    ...[login to view URL] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order

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    Application key features: * high definition audio and video calls * audio c...sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers allowing to reach everyone that has a "classic" phone line.

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    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

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    FreePBX Setup 18 hours left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

    $35 / hr (Avg Bid)
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    Need to set up a Cisco phone to convert to SIP to use with fpbx. Only person with direct experience in it need to bid.

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    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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    i need a software to remove or change SIP RTP PayLoad like if i install this software on os it have to remove or change all incoming and outgoing SIP RTP PayLoad if you can make it please let me know

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    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    Sip and paint artist needed for one week event

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    Looking for an On the Call Technician, who has some ex...6. Basic Computer Knowledge 7. Have a laptop or desktop with windows 7 or higher. 8. Photo ID 9. Paypal account for payments 10. Headset Optional but preferred. for windows sip phone, or android, iphone 11. willing to be creative, and outside the box thinker 12. Honest Contact me to Learn More

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    Hi Dear In running local FreePbx server under elastix and the server have 2 network card One card link to Sip line from local Telecome company and another network card for internet connection My local server is online now via port forward from the router and it’s working by Zoiper app very fine in computer an mobile... in local network and online

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    Hi Dear In running local FreePbx server under elastix and the server link to Sip line from local Telecome company and there is another network card for internet connection My local server is online now via port forward from the router and it’s working with Zoiper app very fine in computer an mobile... But the problem its just working with Zoiper nothing

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    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or phonegap or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through a SIP proxy. we will provide all the server details needed

    $1797 (Avg Bid)
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    Description. A simple PBX pure software would have to be [login to view URL] software should be installed on Windows Caller calls the number to Will and can select appropriate department through IVR. The language should be stored under MP3. Customer lists to maintenance music + and waiting position. Once customer is connected with the agent and the agent wants

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    i need to configure 3cx and add sip trunks

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    I want to add new SIP domain in skype for business and make it working.

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    ...Speex (narrow/HD/Ultra HD), SILK, L16 PCM (narrow/HD/Ultra HD/CD) Data Flash Management (Multi-Partition File System, Advanced Firmware Management, Bootloader, Secure Area) SIP Fusion TCP/IPv4/IPv6 DHCP Client Extensible Command Line Interface Web Server with Extensible Configuration Web App and Configuration and Control Web Services VoIP Call Manager

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    ...Today our fans want to ask you a few questions about some of your paintings and the media you have used. Daumier: Well of course. Daumier picks up his water and takes a small sip. Presenter: The first painting our viewers are interested in, is your print Gargantua. Daumier clears his throat. Daumier: Ah yes, one of my finest caricatures. Presenter:

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    ...general familiarity with backend frameworks likes Rails, Django, Phalcon, YII, etc. Fluency in SQL Databases, NoSQL Databases, Cloud Storage APIs, etc. Fluency in TCP, XML, SIP and other web protocols & patterns. Experience with building secure web applications, troubleshooting, well scalable applications, etc. Extensive experience with API architecture

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    Dears Developers We are ruing Sip service under FreePBX platform And we use [login to view URL] service to keep our server online using router port forward . So we need app with SIP protocol support to make viop calls with our Labled & Customization with Full Branding Mobile Dialer support IOS iPhone and android mobile. Mobile Dialer FEATURES 1-User friendly

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    ...hire new people for the NOC, so ongoing training is important to us. Here are some suggestions as to what the training should cover. We want to focus on Voip and SIP. Why Voip SIP - protocol structure, format, Commands, how to read it, Error codes, interpreting the error codes. Under Voip introduction to Voip Why use Voip over Analog How

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    the scenario; - I have sip phones registered to my ip pbx - when a sip phone try to call a mobile/landline - my pbx will route the call to "skype gateway" - skype gateway will use a skype account to connect the call transparently sip phone -> ippbx -> skypegateway -> gsm/landline inbound calls to be handled with ip pbx the project should be a windows

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    ...Android APP that works as a SIP Intercom... A big botton on the main screen the show (CLIC TO CALL ) when clic it will make a VOIP (SIP) call to an espesifi extencion, also if I call the extension it needs to autoanswer. on the config menu I need to be Avaliable to change this values Sip server IP: Sip Port: Sip User name: Sip Password: Number to c...

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    I need a C...multi-tenant like Bicom PBXware where I can create virtual pbx to resell to my customers. My customers can access his accounts and manage IVR (with easy gui), configure trunks, sip accounts, registrations, hear registrations & billing and so on... Bicom PBXware software Link : [login to view URL]

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    ...The features/ pages would include 70-80 Pages around. NAV of MF linking for fetching Price (NAV) info (we already have the membership of BSE). Retirement Planning Calculator SIP Calculator other calculators with various drop down Home page: Scroller images Corner: Login call us icon

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    Sip tool that will send at least 1000 calls.

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    Use a raspberry PI 3 as a PBX/IVR 1) PBX system (FreePBX?) with SIP for google voice (& callcentric) 2) 1-3 extension to go to a rj11 jack (preferably 3 if possible) 3) 1 extension to send a signal from the pi to hardware specified by the dev to close a circut, only after entering code (for example when selecting extension 9, it should prompt for a

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    Looking for someone to build me a flooding tool (for testing purposes only). The tool (software, app, script, etc...) needs to flood any United States number with at least 100 calls per minute. (500-1000 per min will be better). Every call must come from a different number (different caller id) either from a list of numbers I will upload or preferred is I will input the area code to the tool (for ...

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    Add to feature Proxy and Ath. Name

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    Would like to have a design for a on-boarding platform. Similar to the following link. [login to view URL] Looking to out do this process.

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    Hello Dear i have Opportunity to work together. now i have local FreePBX system using SIP line with 5 trank and 20 extension from our local telecom company STC. And i use this system for CallCenter serivce by Zoiper application, and i have agent work on it in my office. so I want connect the local FreePBX system to anther online system to we can allowd

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    Hello Dear i have Opportunity to work together. now i have local FreePBX system using SIP line with 5 trank and 20 extension from our local telecom company STC. And i use this system for CallCenter serivce by Zoiper application, and i have agent work on it in my office. so I want connect the local FreePBX system to anther online system to we can allowd

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    I am looking for an outbound sales team that is capable of each making 100 calls per day. Phone service/SIP Server will be provided. There’s base pay plus commission opportunities. You will be soliciting local businesses of our telephone services. It is required to speak clear English, but this job is definitely not limited to the United States

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    Add sip input tab to frontend to phone dialer.

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    Dear Amrit M 1-i have FreePBX system on -Virtual machine CentsOS 5.10 . 2- you can login via teamViwer on widows7 that the virtual machine on it and then you can login t...the virtual machine on it and then you can login to cents vis SSH or via VMware 3- there is web controal also for OS and FreePBX 4- we just need to add new setting for new SIP line

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    Hi Sunil I saw that you have experience with SIP and VoIP, on android and iOS I need someone to build iOS and android App. If you are interested we can continue talking on chat See you

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    We have a few Skype Connect profiles with inbound numbers at...script should (persistent) connect to Skype servers, login and answer the calls by returning a recording (depending on the inbound number). That's it folks:-) It's all standard SIP, pretty straight forward. --> $100 FIXED (it's max 3 hours work, MAX, 30 minutes if you've done it before)

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [login to view URL] i want to run this program on a router Router Link [login to view URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Help programming sip endpoints, applications voice & sms in [login to view URL] I need to connect my [login to view URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [login to view URL] Prior experience

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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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