We are porting our existing SIP applications from using the Radvision SIP stack to using PJ SIP. Because of this we are looking for an experienced C++ developers that has worked with PJ SIP. The company is based in Phoenix, AZ, U.S.A so we would prefer a developer that is within the U.S. Any applicant must have at least 5 years using C++. The applicant must know SIP and work in-depth with PJ SIP. Phoenixsoft develops a soft-switch that provides, prepaid, Class4, and Class5 services. There are three application that make up with SIP interface. These are our SIP Proxy, SIP Registra, and the SIP Call Processing layers. All the application run on RHEL/CentOS 7 and are multi-threaded. It is estimated thi...
android and IOS app, the user login using a user and password, sip server has to be hardcoded in the app, once he login a "call Us" button should be there when he presses it just call a hardcoded number on that button, no need for the key bad on the main screen only after the call initiated the key bad is needed except log out, our logo will be provided below is a picture for what am thinking about. the APP's should be uploaded to my IOS and android store.
I am looking for someone to assist me to build a web based interface using React for a softphone used to establish voice calls through a browser. The size of this should be responsive. Width is generally fixed, estimated to be about 200px, height is about 400px but may span the length of the entire page. Elements should be centered. It does not have to be functional in the sense that it does not need to be able to actually make the calls. Instead, I am just looking for a frontend interface with some placeholders that will allow for javascript functions to be added in at a later time to perform the necessary functions (i.e. perform the actual call). Essentially, the softphone would look something like the dialer on an iPhone and provides transitions across various different ...
Configuration of Agent Softphone on an Installation of Queuemetrics reporting running on freepbx server Installation of the components already completed and running....The only remaining aspect is configuration of the softphone due to the https aspect for tomcat. We require someone with adequate knowlegde of the subsystems to successfully configure the Softphone on the Agent Wall as well as on the agent page.
We are a new company in the UK building garden office pods. We need an architect to draw up 2D and 3D CAD designs based on hand drawings. The designs will be for SIP board construction and will need to be heavily detailed listing all the materials, size and measurements for the manufacturers and construction. The work will be ongoing for the wining bid. We will need to see example work before any bid is accepted. Design examples and a full set of requirements will be provided.
FreePBX V with Twillio SIP trunk using PJSIP driver. Calls drop after 30s, narrowed down to not receiving an ACK in response to initial OK after INVITE on incoming calls. Require diagnostic of the reason why the ACK is not being received (Twillio PCAP suggests that ACK is not being sent at all) Suspect issue is related to Contact Header IP not being using properly. Will provide remote access to Freepbx Gui, Commandline, and Twillio Console remotely. PCAP will be provided to suitable bidders. Needed ASAP on business critical line.
-Need a system to convert incoming whatsup calls to SIP destinations. -Need a software to make and receive WhatsUP call from laptop (without the use of the phone active).
I have a grandstream model UCM 65 10 that reboots and connects the voip trunks very well. however after some minutes, the registration refresh is not working properly. need help: this is the error message in the asterisk logs WARNING[1923] chan_sip.c: Timeout on 896482223-292013357-1450912169 on non-critical invite transaction.
We need to setup a ITSP phone number to BigBlueButton conference bridge, the purpose of this is users can join webconference meeting from their Mobile phone(call). For this we need to configure FreeSWITCH to receive incoming calls via session initiation protocol (SIP) from our nexmo(ITSP) provider. We are looking for someone who has deep knowledge in freeSwitch and BigBlueButton.
I am looking for a freelancer able provide a system to do Voice Broadcasting System. It should able integrate with SIP Phone to enable to do call out.
I need a professional who has experience in PFSense to create a configuration for forwarding to internal servers according to the external domain. We value information security and would like access to PFSense to be internal only. We also use SIP channels, we need to pass these channels through PFSense. If possible, we would also like an installation of a network and server monitoring tool, but this is not a priority.
I have odoo v11 community edition. I am looking for a freelancer who can help me to implement VOIP calls form Odoo. I would like to implement function of click to call in contacts.... I dont have any asterix, freePBX or cloud calls system yet, I do have SIP Trunk lines. I have seen many projects that had integrated odoo with FreePBX system, and I would like to do the same. So in short you will be connecting odoo with our freepbx system. the odoo crm, contact, leads, customers and more, should have click to call function, popup for outgoing call. If you know odoo 11 and you already develop/integrate phone system, please let me know. I need this job to be done ASAP. Thanks
I am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr wit... to build me a carrier grade platform for wholesale and retail. I can work with cdr without rates for now. Platform must be able to handle thousands of concurrent calls and also doing rtp proxy. Platform must have a did management feature and the ability to provisioning and turn up customers immediately, retail with1 number or wholessle with thousands of numbers. Cdr must be very detailed with sip trace available with the click of a mouse. Basically all the features of the latest version of opensips EXPERTS ONLY. If you have a customized wholesale version of a2b I am willing to look at it also. Maybe doing signaling only with a opensips proxy in f...
Hello, I need a FortiGate VOIP Expert to take a look at my firewall. I use a NON FORTINET VOIP SIP server and voice quality is horrible. Thanks.
Hello, I'm running VitalPBX. Using PJSIP extensions with SIP trunks. All with TLS signalling and not encrypted RTP. When placing multiple calls at once, voice quality is horrible and choppy. I need someone who can look into Asterisk settings for me and possibly do a network capture and understand what is going on.
Configuración siptrunk 3CX proveedor CallmyWay Costa Rica Necesitamos configurar el sip trunk hoy
********please ONLY Adobe Connect Admins and EXPERTS send me message!!!******** we need create SIP phone pods to adobe connect hosted (on promise) server. *******you must show me worked phone connected room before buy services.********
Need someone to configure click2dial to make calls. Calls using softphone and freepbx - done. Odoo is connected to freepbx - done. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. Thanks
I tried many ways to get US, UK , Australian virtual numbers to Oman but I am unable to get any incoming calls .. Need working techniques to get the calls here without using VPN . I already have US google voice number but I am unable to get incoming calls with that as voip is not supported here Need working technique from tech geness
We have a newly installed FreePBX server hosted by Cyberlynk. The PBX was setup by our SIP trunk provider. Currently the UniFi phones receive calls but with no audio. The UniFi phones send audio to the other caller fine. We need someone who has experience with Hosted FreePBX AND UniFi VoIP phones to assist with resolving this audio issue. There's not much documentation on the UniFi phones and they are difficult to work with so please only apply if you have this exact experience and can resolve our issue.
We are looking for a programmer with experience in working with audio/video (A/V) SIP VoIP communications using WebRTC and WebRTC mobile, call managed through a PBX (preferably Asterisks). The first and main engineering task is to establish A/V SIP VoIP communications from a Pine64 SBC running Linux Armbian through a Web Browser to a smartphone (iOS/Android). Asterisks and the WebRTC services are to exist in the Cloud. After the successful completion of the above task, we would also like the to establish a connection from the Pine64 to a windows 10 PC (running the Chrome web browser). The objective of this project is to create a reliable communications path accessible to other developers through API calls at each point. These other developers would develop Web/Node.js, iOS, A...
We need to find a coder to create a cross platform softphone (Like Bria from counterpath: ). This should be done with Electron/nodejs, with installation packages for Mac / Windows and Linux platform. The project involves knowledge of SIP protocol and VOIP concepts.
Hi We have installed vital pbx 3.0 and we are looking at moving all our clients over from kiero operator to vital pbx. I'm looking for someone to move the clients over one by one. I will advise you the call flow setup along with a zip file for each client for IRV's, welco...Tennant in vital pbx Along with configuring other vital pbx settings for clients to get the most out of the platform. We as a business will be the first customer to go live on the new pbx. Once we test the platform for a week then other clients trunks can be added and they can go live We have around 100 extention across all clients With your experiance we will also look at other wholesale sip providers to save the most amount of money and provide redundancy for us. We can discuss either an h...
Hello, do you have experience to work Rockchip ARM-Cortex A35 and Android 9.0. It must work Android UI with 11r and SIP application. So, it is IP-phone.
We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single message then the user will be forced hangup.
...use both: firestore implementation suggested by google codelab or a homemade webrtc signaling server coded in nodejs (we have a signaling server wrote in nodejs already running and usable) Tech informations about protocols (webrtc, sip) will be provided on job accepted. Actually we have a working demo environment based on core and critical functions/protocols needed. This a “step 2” implementation to test it on a real case. Codelab used for our demo environment can be found here: rtc-codelab We have a perfectly working sip/webrtc phone server on our DMZ ready for our porpuses....
Build audio and video conferencing site with webrtc and sip using jitsi. Front and back end with good management system.I am looking honest qualified developer who wants to work long term to combine these 2 sites. and blogtalkradio.com. with video conferencing part of and some features from .there are very simple but i need honest reliable people i can work with. NO PREPAY OR MILESTONES. Milestones are total garbage. 99.99% of a project is 0 to me . So if you are qualified and NOT a crook. Lets talk. Your bid is the final price. I will not pay a penny more.
...to be changed as required. I'm sure if you've read this far you know whats required in order to build this. We want to make use of VoIP/SIP Trunks to send SMS as we were doing previously with another business partner. Each SIP trunk could send approc 12,000 SMS per hour which we want to replicate with this. SIP trunk costs around 20 USD per month per trunk, which you need to source. The SIP trunks must be 'pooled' rather than allocated to each account, this way they are maximised across all clients. Longcodes required for each campaign rather than per account, again you are required to source these also. ideally, we need system we can take over and order SIP trunks to system and longcodes as rrequired. Clients to have a 'sendin...
...application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Convert audio from/to SIP and GSM networks - Able to run on Andro...
...application that runs on Android phones (4.0 or above) and can receive calls through SIP and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requirements are unable to be met, please let me know before we make a deal. General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Convert audio from/to SIP and GSM networks - Able to run on Andro...
I need to turn an Android cellphone as SIP to GSM Bridge of Gateway. Traffic from SIP must be translated to GSM channel. You are free to choose how to implement this feature. All that is required is that in the end the Android phone will be acting as a SIP to GSM channel/trunk/GW and other SIP devices can connect and make phone calls using this. From my side i can provide you with Linux firmware and documentation for few native SIP > GSM GWs. This is the first thing which is need to be developed. Depends on result of a job contract can be signed to future cooperation.
I have an asterisk server which has oem set functionality which I want to be accessible via an android app. Below are the requirements. Allow users to be able to call each other within the netowrk via UserID - users must be able to call each other at anytime i.e. constantly connected to the SIP Include speech synthesis in the app - so user can type in what they want said at the start of the call and it gets sent to server Ability to set Caller ID internationally View call history Set voice pitch dynamically through app if possible Receive DTMF dial tones as sent to server by callee Must be modern design and run smoothly. Must include FAQ with details on how to use the app Must include help page with contact information On startup: must include a progress bar with audio file play...
Hi Eremin P., I noticed your profile and would like to offer you my project. We can discuss any details over chat. We need to make Kamailio to work with MS Teams and a SIP provider. Part of the configuration is already done, need to finish the number rewriting, RTP proxy and routing part. (Teams is already connected)
Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.
Hello Folks, I am looking for Linphone expert. We are done with entire customisation but we are facing issue in call. Below is exact issue Apple requires the use of PUSHKIT to receive silent notifications when the app falls into the backgr...am looking for Linphone expert. We are done with entire customisation but we are facing issue in call. Below is exact issue Apple requires the use of PUSHKIT to receive silent notifications when the app falls into the background and the app is sleeping. PUSHKIT requires an IMMEDIATE call to CALLKIT which starts ringing the call. The call is now ringing without the SIP invite having arrived at the app. The app registers to the SIP proxy after calling CALLKIT. The INVITE is send AFTER CALLKIT starts ringing the app. Please bid if you ca...
Title: Sip or Spill - A Bachelorette Party Drinking Game NO AUTHOR Dimensions: 6 x 9 inches Needed Final Formats: PDF, PNG, JPG (RGB & at least 300 DPI), and Ai/Vector + Layers/Fonts (You can zip the final files). For the title, I would like the font to be bold, but girly. I have attached two sketches of some ideas I had for the cover, but I would like to see what you come up with yourself. The font should be clear and easy to read with no shadowing affects. The text should be clearly contrasted against its background. Please use cohesive colors that blend well, it should be eye-catching and vibrant but not distracting or too busy. I want the big illustrations and the title to be the main focus, the background should not be super busy. Do not make the background white. The ill...
Ciao Raffaele, ho visto il tuo profilo e da quanto letto suppongo ci siano i presupposti per sottoporti un lavoro, anche con la semplificazione della lingua, che dovrebbe essere per noi un concept da discutere col cliente per capire se renderlo produttivo o meno entro 1 mese. Le tecnologie che verranno toccate saranno principalmente webrtc, js e sip.
my project is simple, Install and configure stable RestComm smsc gateway ( ), so smpp to sip message should works with good performance. Centos os is preferred but you can work any os.
...want a website where people can register with the following: Name City Mobile Number Email PAN Number Date of Birth Once registered, they can log in and be able to see: a. List of all Mutual Funds Units he is holding in India b. Current Unit Value of the Funds he is holding c. The net worth of his Mutual Fund portfolio d. P/L for each holding Further, he can filter the above info on "All" or "SIP" or "SWP" or "Lumpsum" etc. with respect to payment mode. We do NOT need any units purchase feature or reedeming of units. Just viewing the details. Clearly mention the following in your bid: 1. What technology would you use? 2. Which API would you use? 3. What would be the cost of the API? 4. What would be your development cost? 5. Exac...
I need to connect webrtc softphone to freepbx extensions .I only need to make phone calls between extensions
We need drawings made for a 3 bedroom 2 bath SIP home.
-Install FreePBX on a LAMP server and configure trunks.
i want to make a SIP dialer dialer have to work all network codec must be g729
Set up Kamailio in front of asterisk server to proxy webrtc/tls /tcp / udp to asterisk server on private network with failover. - rate limiting sip traffic by source IP - dropping malicious / invalid packets - integrated with APIBAN () - MySQL integration to be able to adjust config (like primary / secondary / tertiary failover etc) - auto install script so this can be reinstalled / moved / etc easily with minimum effort. - Bonus for early start / completion within 24 hours
We need a phone vendor for a SIP migration. Customer has existing POTS lines and they are migrating to ATT IP Flex connection.
1. Linphone registration a) Linphone sends REGISTER message containing its PUSH token via Kamailio to the PBX b) When Kamailio receives the OK from the PBX regarding the successful registration it stores the Linphone push token and username Incoming INVITE to Linphone PBX---->Kamailio a) Kamai...INVITE from the PBX and does not forward it to Linphone b) Kamailio sends instead a PUSH to Linphone to FCM (google firebase) c) When Linphone receives the PUSH it issues a REGISTER message d) Kamailio catches the current contact address from the REGISTER message e Kamailio now forwards the INVITE from step b) to the current contact address of Linphone f) The usual INVITE from step a) proceeds SIP-INVITE ----->[PBX]----->[Kamailio]push------>(android[UA/SIP_REG-PBX]-...
Looking to build a setup where. 1.) UA /WEBRTC client registers to kamailio 2.) Kamailio is registering directly to PBX for all SIP. extensions. 3.) setup where Kamailio doesn't use RTP engine (not sure if i can) need to test config is almost there Need some assistance/tweaking. as i dont know.
We are looking to hire a full-time help desk tech for our IT Consulting company Job Duties - Windows Desktop Support - Agent / AV installations - Remotely help end-users with Printer / Software issues - Configuration of new devices including Desktop / Cell Phones / Laptops / Tables The tech will be ...are looking to hire a full-time help desk tech for our IT Consulting company Job Duties - Windows Desktop Support - Agent / AV installations - Remotely help end-users with Printer / Software issues - Configuration of new devices including Desktop / Cell Phones / Laptops / Tables The tech will be providing with an office 365 account / RMM Account to remotely connect to computers / and a softphone via Teams to place & take phone calls Decent but not perfect English is ...
hi, We need a App that can make and receive calls through sip client. development should be in xamarin. thank you
Seeking for someone who can help with a sip integration. Please only people with experience in VoIP and / or jitsi. Thanks
