Blackberry sip softphone jobs

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.

********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...

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

Hello, We are looking for someone on an urgent basis who has full knowledge of Blackberry UEM configuration. We would need you to do the following: Want a proof off concept configured and brought live android ios 2 Profiles to choose from each OS Want someone with knowledge of VPN, secure connect and dual dar Want everything setup including the app store and app store application publishing linked with profile No corporate apps to be pushed all apps available from app store Want a quick configuration put together so we can start testing. Please let us know the following 1. You experience with UEM configuration 2. Hourly Rates Thank you!

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

my cafe name is "Eat 'N' Sip".intrested freelancer can bid on it.

I am Sathish from INDIA. Like many am one of the Blackberry fans out there in the world. I couldn't accept the fact that Blackberry OS no longer support any applications via BB World. I always wanted to look for the options to bring back those applications again. Recent times blackberry os accepting android .apk files and sideloaded .bar files. I do see there few apps developed and its working pretty fine, but if we improvise the same and we can bring back this legacy So I am looking for Blackberry app developers who can develop mobile application in .bar file format. Also you can suggest me and advice me with any ideas/opinion. P.S: Charges are negotiable

We're looking for a Shopify Design expert to design our new Shopify store for a Paint and Sip studio. This will also include GuestManager Shopify plug-in as the Event Ticketing platform, which will also need designing (eg. Ticket design). We are looking for someone with extensive graphic design experience that can also work with the Shopify theme to produce the live site.

Hi there I have a Softswitch name ASTPP V3.6 which is running on freeswitch There is failover gateway options which I try to use it but its not working want solve that. And also one more issue that sometime SIP account in unregistered automatically. Thank you

Our Kamailio server configuration is not working with sip registration. Need reinstall of the Kamailio server. Please apply only if you have already worked on Kamailio

Hi As per description, I need 1x UI Back end (Dispatch ) with our logo hosted on our server and domain 2 x Androids Mobile Apps (1 for the Driver and 1 for the Passenger) 2 x ios Mobile Apps (1 for the Driver and 1 for the Passenger) The Companies Colors are RED, White and Black THIS IS OUR CURRENT APP WANT SOMETHING WITH SIP OPTION ADDED FOR A SOS BUTTON FOR EMERGENCIES AND PAYFAST PAYMENT GATEWAY.

Webphone for integrate to our webbase sales management system. Functions : Answer, Hold, Transfer, Reject, Mute API Strong Installation Documantation (Word, Video)

I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory

...Phone. The target uses ARM architecture, runs Pine64, an embedded platform and has Node.js based API running on Linux. Deliverables for the POC: We would like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsite APP communications. Establish a webRTC to SIP(PSTN) connection having two-way audio call and local VoIP connections (Guard Phone). If an alternate viable method is proposed we can consider it...

Hello, US accent agents wanted. Hiring worldwide for remote agent support. Must have a good internet connection and a softphone with phone number we can forward calls to. Please feel free to provide a link to a voice sample. To start you will log in to our answering service and return calls. You will have the opportunity to ask any questions about your response to the call/s prior to any call backs. Once you are trained and understand the service and are suitted with the conversations you will be able to take live calls. Base pay plus commission. We are a generous company and the services you will be answering questions to will be between 500-2500 US. Commission percentages will be based on sales and performance. We do not want you to have a "sales" personality....

Need help integrating DR3000 repeater with SiP based system

hello Desginer We'd like to design 2 banners for Collagen Sip Straw with following contents ( key message) 1. Radiant Beauty with Collagen Sip Straw from Japan for 4 weeks 2. First Collagen Straw in The world Reference: - - Thank you