Blackberry sip softphone jobs
This is to call Spain customers to ask for feedback on their recent purchase and offer a discount code on repeat purchase. There will be hundreds or even thousands of customers to call. You need to have good connection to Internet and use softphone APP to make the calls. We will provide the resources. Previous customer service experience is preferred but not a must. We pay by number of calls completed.
My Blackberry Bold 9930 crashed and I've got a white screen - App Error 200.... It will not boot up. I've got a backup from June 14th that can only be accessed from the device. I need to retrieve the backup from the crashed device so that I can restore from that backup to a new device.
Hello Want to develop API fot nat traversal. My client is behind nat firewall. Client (softphone) want to connect to sip server by resolving nat as well as dynamic ip address.
SIP SoftPhone for Salesforce Salesforce integration with the Mizu WebPhone
My Blackberry Bold 9930 crashed and I've got a white screen - App Error 200.... It will not boot up. I've got a backup from June 14th that can only be accessed from the device. I need to retrieve the backup from the crashed device so that I can restore from that backup to a new device.
...while the call between the external number and the ext 100 continues. In some types of failure, a graceful error would be nice. The second internal extension should get the original caller id information passed through as is. I will handle the crm integration. When the script is called, i can pass the ip address of the user calling the script. The script will have run a asterisk cli command (sip show peers or similar) to retrieve the extension and or live call associated with the ip address. If there is a asterisk config file modification needed, the change must persist changes made by the freepbx gui. Part 2: I need a method to transfer a call to a wav file through the asterisk cli or other simple method. The intention is call the asterisk CLI from a simple php exec s...
professional with great experience in Asterisk, PABX, predictive dialer, softphone and wertc
I am not strong on Python and need help (share screen and give control to VM) Environment: Ubuntu 20 LTS, Python 3.8, Asterisk v16 Note: 1. Asterisk (works OK - Inbound call, plays monkeys via a context with an external softphone) GOAL: 1. A "hello world" python to play the "monkeys" prompt on an Inbound call using Asterisk ARI REST API. 1. Using this Python module ARI 2. See "Hello world example" My basic issue/problem >>> import ari Traceback (most recent call last): File "<stdin>", line 1, in <module> File "<frozen zipimport>", line 259, in load_module File "/usr/local/lib/python3.8/dist-packages/", line 8, in <module> File
Hi there, We need a service of configuring SIP conference bridge for BigBlueButton. Our server is behind NAT, and we can't reach the result - system always says incorrect pin. Would be grateful if you can correct it.
Hello , We need Android and iOS Sip Mobile Dialer with Integrated Tunnel/Bytesaver to work in blocked country like UAE Middle East .
Dear developers looking for a developer who has already developed or have good experience on VoIP sip mobile dialer and did .we need a mobile app where people can buy DID numbers and then use them as 2nd phone and make international calls on cheap rate through that app
I do have a small VPS configured with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures: with the FreePBX Asterisk distribution version FreePBX that I have problems finishing to configure as incoming calls from my trunk seems to be bounced as you can see in the following pictures:
Receiving a SIP/2.0 420 Bad Extension from Kamailio on register keepalive The first register is OK. After 120 second i got a 420 Bad Extension. ******************** REGISTER sip: SIP/2.0 Via: SIP/2.0/WSS ;branch=z9hG4bKFNPKsWQBArkN7rQwDnIuoH4p1ZCSJS8q;rport From: <sip:991125943220104@>;tag=qMzPBHwxnsxK3XAQVepG To: <sip:991125943220104@> Contact: <sips:991125943220104@;rtcweb-breaker=yes;transport=wss>;expires=180 Call-ID: fcde2b26-a2cb-7809-1606-7f7183736053 CSeq: 45151 REGISTER Content-Length: 0 Max-Forwards: 70 Authorization: Digest username="991125943220104",realm="",nonce="xxxxxx",uri="sip:",response="xxxxxx",algorithm=MD5 Organization: XXX User-Agent: XXX webRTC UA 1.0. SIP...
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.
...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. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice versa 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 - Route call fr...
This is is to install and configure FreePBX on a VM. Below is the list of objectives. - Install FreePBX on a VM - Configure network setting and activate. - Connect to Yeastar S100 IP PBX for outbound calls. - Help to configure an extension. - Help Setup an IP Softphone (any free softphone) and make an outbound call. If you have any questions please ask me.
...necessary guide on the system installation,configuration and usage. In summary after the configuration Vtiger users must be able to make outgoing calls, receive calls, and have access to recorded calls. Note: users in crm & freepbx soft phone extensions are already made. Only linking between the 2 servers has to been done in such a way that every crm user's call is routed through the particular softphone extension which is assigned to that user. Also while implementing click-to-call small change needs to be made in the crm which is that the mobile phone field should be masked, last 4 digits of the mobile phone should not be visible to the crm user to increase data security....
OWN HOSTED CLOUD SERVER WITH IP-IBX SYS WITH CUSTOM FEATURE , SET CALL ID , CALL OUT , CREATE USERS , KNOWLEDGE OF SIP TRUNKING
Looking for an experienced twilio person to help with configuring a new sip server on twilio to work with Linphone softphone. Immediate work
Need to build a server client application for SIP TLS and RTP MUX between 2 systems, whereby 1 box is behind NAT. TLS transport layer security will be used to encrypt SIP signaling and RTP MUX – RTP content for concurrent calls between 2 peers are transmitted in a single UDP packet.
Telephony mobile app for calling Card customers with a superior experience using iOS/Android. Also make use of APIs to communicate with a web, telephony and other 3rd party platforms. Unique and critical features it will possess include an embedded softphone WebRTC(prefered) or SIP based and mobile dialer.
i have my sip server on france now i want to register sip from a blockage country like UAE UAE have all voip blocked i dont want to use a separate VPN i saw lots of dialer there have proxy server and its work over dialer proxy so i need you to make something like this we will use a middle server like a proxy server like we try to register on our proxy server and its FWD to all my SIP server or we can make sip proxy or anything as you like i just want to register sip and make call from blockage country if you can make it please bid
Responsibilities Handling Linux Server Install Switches, PBX on Servers Maintain VoIP Billing Providing Support Server Error Handling Write Custom Codes for Development Requirements Minimum 1 Year experience in VoIP Good Knowledge of Freeswitch and Asterisk - working on astpp billing, fusionpbx etc. Good knowledge of mobile dialers an advantage too Well knowledge of SIP Servers Good in development of AGI,FastAGI, Dialplan, FS Scripts Knowledge of JavaScripts and Lua. Working Knowledge of LAMP Full time online position Educational Qualification - MCA, B-Tech,
Two 12 year old girls have started a cosmetic company. It is called: Blackberry Beauty They want a logo for the company. They will initially be making lip glosses and perfume. They like the colors purple, blue, and black. But it doesn't have to have all of those colors or any of them.
hello I will offer remote desktop to configure a freepbx asterisk install. you have to add a second ntw card , do static routing and integrate sip trunk with my telephony provider
Hello, I want a developer who can create a SIP investment calculator for me. Whether it's a plugin or html page that can be embedded to Wordpress. The plugin/page will just have to calculate that If I invest "X" amount of money for "Y" amount of year and expect "Z%" return then how much will the person have that time. If you need some samples, please PM me. If you are up for the job, please PM me with your price. Regards
I need a fully functioning Softphone app with branding and rebranding functionality. White labeling and reselling features built in to securely allow the hosting. VoiP SIP
hi, i need a general price and fees associated with designing a fully functioning mobile voip/ softphone/ SIP app. The app needs to be free standing and built from the ground up. And needs to be coded to allow whitelabeling the app to other suppliers as well.
I currently have ISDN PRI setup on voice gateway cisco 2951 and CCUM 10.5 with about 50 DIDs (numbers) I have new setup: ITSP SIP trunk to claritytel on my voice gateway and another trunk to CCUM. This is just additional new config to ISDN PRI - existing one. I have user and password from Claritytel. Besides creating both sip trunks we need to create other config so for example using digit 7 in front of any number the SIP trunk is dialed (not existing ISDN PRI trunk) I can provide more information as needed as well as detailed plan. This needs to be completed so I can use that SIP trunk from any Cisco phone in the network.
1. Make a three picture facebook header: I uploaded a sample from a surf, skate and wake event AND one example for this project. Use a pedal pic, a group shot and full boat pic. Text should read: Coming Soon! 2. Do 3 individual COMING SOON social media pics using three different pics. 3. Combine the Sip-n-Cycle Pedal Cruise logo, the pedaling video and the paddle video and a group pic with the text COMING SOON. This will be a short video. Add upbeat music to the background.
setup the SIP on freePBX and asterisk IVR managing call routes etc
online e-commerce website specialist on selling coffee beans and coffee equiepments. need a good looking and creative logo design. along with the logo I need two things bag design and shipment box design. also, I need the logo to have the name on both Arabic and English name of the business: English: First Sip Arabic: اول رشفة i uploaded 2 design examples ideas
Configure Elastic PBX on tls for sip calls and on untangle port range forwarding
We have a project with Huawei equipment (Core Switch - Access Switch - IP Telephone System) Currently Status: we have all 4 land lines configured on AT0-slot4 with designed prefix 6890316, but this allowed to use any one from 4 lines for any telephone...Switch - IP Telephone System) Currently Status: we have all 4 land lines configured on AT0-slot4 with designed prefix 6890316, but this allowed to use any one from 4 lines for any telephone which can cause to use all lines by one building and hold other building connection till have free line. "Project has 4 buildings" Required Task: Need to assign outside land lines to use by dedicated IP-Tel SIP user IDs as per below distribution: Port 0, 2 : Telephone ID 200 - 226 Port 1 : Telephone ID 100 - 103 Port 3 : Telep...
...send/receive calls through SIP (Can use any sip stack like sipdroid) 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. Summary The goal of this project is to develop an Android application that can send calls through SIP and forward them to the GSM network AND receive calls from GSM and forward to SIP. The application should then forward the audio and convert from VoIP to GSM and vice vers General Deliveries The application working in APK format Full source code Simple manual for compiling and generating the...
In Bank transfer module I want to add a transfer sip uploader which i can see from admin panel
Hello I need to do a direct inward dialing system is a call system to receive calls you have to receive calls on softphone (extension) for example I need a virtual US number. USA With 10-channel DID for incoming service I just wanted 1 virtual US number. USA 10 channels means ... If there is any incoming call on that virtual number, it should appear on 10 PCs by softphone at the same time. To receive calls on that PC, I want sip configuration, I have softphone like eyebeam and xlite For eyebeam configuration I need to put the ID password and Ip domain once I configure that I can receive calls on 10 PCs through eyebeam it has support and can forward all calls on that softphone needs configuration like ID pass and domain IP so if you're able to mak...
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.
Queue Group proper call flow setup -Freepbx Asterisk Project This is a freepbx asterisk project with twilio elastic sip trunking, Everything is setup correctly and working as we wish. Just having difficulties with queue group - Message sometimes your call cannot be completed.' This happens when I try to stress test it with multiple calls simultaneously Need someone who can work on this immedietely and start immedietely so only reply if you are available to start now. Will have more projects after.
We need a simple Voice Autodialer System with call campaign auto scheduling, repeat and call duration features It needs to connect to our wireless gateways via SIP and make calls out form them at a predetermined time and for a predetermined amount of time. Or it can call a play a message of certain length or audio file loop for a certain amount of time. This will be used for network testing purposes.
Task that is already done. Installed Centos os. Installed Asterix server. Task that’s required to do. -Configure SIP trunk and DID number. -Make dial plans for 5 cisco ip phones and integrate with Asterix
A basic version of localphone.com. Build website on our domain to do the following: - Signup or Login to user's account. - Call a new number or a number from account's contacts. - View/manage calls history. - Add/modify contacts (name,number). - Administrator account to delete/block users' accounts. The VoIP call needs to satisfy the following technical information: - VoIP protocol: SIP - Voice supported codecs: G.711µ, G711A, G.729. - Numbering Format: CountryCode, AreaCode, Number. and forwarded to the unique IP of our supplier, to reach the destination of the call.
I have a rewrite condition in my htaccess file to detect mobile device. Google is telling me it is getting a 301 error on iphones when trying to load the page. Can you look at the script and tell me what needs to be fixed? Should take about 10 minutes for someone knowledgeable. RewriteCond %{HTTP_USER_AGENT} android.+mobile|avantgo|bada/|blackberry|blazer|compal|elaine|fennec|hiptop|iemobile|ip(hone|od)|iris|kindle|lge |maemo|midp|mmp|opera m(ob|in)i|palm( os)?|phone|p(ixi|re)/|plucker|pocket|psp|symbian|treo|up.(browser|link)|vodafone|wap|windows (ce|phone)|xda|xiino [NC,OR] RewriteCond %{HTTP_USER_AGENT} ^(1207|6310|6590|3gso|4thp|50[1-6]i|770s|802s|a wa|abac|ac(er|oo|s-)|ai(ko|rn)|al(av|ca|co)|amoi|an(ex|ny|yw)|aptu|ar(ch|go)|as(te|us)|attw|au(di|-m|r |s )|avan|be...
I have a FreePbx application runnig on raspibarry Pi and all the configurations was done, but I have a problem during trunk resgistration (403 error). I think that some configuration, must to be implemented in orther to solve this ...have a router(mikrotik) doing NAT between ISP modem and Raspbary Pi that is running freePbx 2 - all farewall/NAT ruler on router was implemented in order to forward 5060-5062 UDP and 10.000-20000 UDP RTP protocol to FreePbx Ip. (doble check could be necessary) 3 - Cenario: sip server is hosted in Brazil and Frepbx is located in Florida-US 4 - Sip Trunk that is rejecting on freepbx is running ok when is cofigured on Bria softphone on the same NAT that raspberry and Bria device are. In first view, there is no problem between sip serv...
I would like to get developed a website through which my clients or prospect can easily do the following: 1) Make KYC 2) Purchase or Repurchase Mutual Funds Units 3) Start SIP/SWP/STP 4) Can Make their financial plan 5) Can Purchase Insurance online. 6) Can See their portfolios online. 7) Can send us a massage. 8) Can see different schemes or products and their comparisons.
I need help to connect Hubspot to my softphone ( ) for click to dial. I have some API for my ip-phone system
Creating for Telecom1 LLC Sip client app for IOS and android phones. (based on Linphone open source VoIP project)
Develop a SIP VOIP Application. The application should able to work on both Android and Ios Work on both TCP and UDP VOIP protocol Can use any Existing opensource project as a base
