Sip jobs

Need a web browser based soft phone to connect to existing SIP accounts. Would like help configuring one? This can be even be an existing / licensed software. Looking for quick turn around.

i have to jobs , first i want a bandwidth optimizer which can compress bandwidth 60-80% and can be installed on router with USB or PC and also i want to make viber server which can be receive calls from sip server and if the callee number has viber installed on their mobile the call will go through viber , if they doesn't have viber the call should be routed to normal route.

I need Phonegap/Cordova expert. - Required skills & experiences Cordova/Phonegap app & plugin development Angular,Ionic,JQM Parse Push Notification googleanalytics,admob SIP, Asterisk Google map,Geoloacation If you have any experiences in this area, you are welcome. I have various projects in some areas, so I am finding skilled and long term developers. Thank you

i have to jobs , first i want a bandwidth optimizer which can compress bandwidth 60-80% and can be installed on router with USB or PC and also i want to make viber server which can be receive calls from sip server and if the callee number has viber installed on their mobile the call will go through viber , if they doesn't have viber the call should be routed to normal route.

Hi Dear ! we want find sip TLS information from Xfinity connect aplication of Comcast we have user & passworld we need find SIP proxy for use it in other sip phone for make and recieve calls

i want is VOIP Softsiwtch send sip calls and terminate through viber application viber servers i mean. its not a phone application its a server intergration simple asterisk to viber call(prefer is asterisk but it can be any)

Develop a SIP Phone App for iPhone and Android to integrate with A2billing. Similiar to the phone on the link below.

I have 3 Cisco 7940 phone without firmware and the error : "Protocol Application Invalid error". I want to have help to fix that

please follow this sdk i need to create sample py script to do automatic call please make your bid if you expert in python

I need someone to develop an App for both iPhone and Android to integrate with A2billing and Asterisk. The functionalities that I expect of the app are as follows: - Show Credit Balance - Ability to make Calls (Off Course) - Show Call History - Show Payment History - Change Password - Ability to Recharge using either a credit card or Paypal - SMS (looking to add an SMS gateway) - Chat (this is needed only be to chat with the support/helpdesk team) Other functionality will be for a local address book within the app. The ability for users to set up favorites (people, family and friends that they call often). Our company logo will be displayed on the app. I am looking for something that is quite easy to use, and also visually clean looking. In other words, not to complicated. Dra...

We looking for Android developer with previous experience in mobile app development for internet messaging/SIP voice and video communication. If you cannot demonstrate your previous work, please do not bid as your bid will not be taken into consideration. More details will be given after we chosen candidate.

I need to describe the product using a commercial perspective, I have all the information in a word document, I just need commercial words. The info I need for the website is: Home: 1. About the product 2. FAQ 3. Partner ...the AES algorithm (i.e., 128, 192 and 256) are sufficient to protect classified information up to the SECRET level.” This modern cypher system allows you to organize your own encrypted communications network with no limitations on the number of users and with support for multiple user conference rooms. Cellcryptor features: •Free calls between users •International calls at low cost using SIP Trunking for secure calling. •Possibility of using virtual telephone numbers for international incoming calls. •Implementation of b...

i need some one who has expertise to setup SIP client on iphone, i already bought sip account from s0h066,co,uk just need help to configure it on my phone thanks

We need to develop a SIP phone bar (web) with receive/make call function and event notifications (from our API) Can be based on HTML5/SIPML or flash

Develop a SIP Phone App for iPhone and Android to integrate with A2billing in two days! If you already have the source code available only then we would be interested.

...reference list. - At least 23 Academic journal, article must be acknowledged by use of accurate Harvard-style in-text referencing; research will be evidenced by inclusion of a reference list. -Compare 2 countries with different industries: Japan’s Toyota (Automobile industry) and China’s Huawei (Telecommunication industry). -Using theories of Dunning’s OLI model, Matthews’ LLL (Triple L) and SIP (Strategic Intent Perspective) theory by Prahalad model to discuss the examples. Details and structure will be further discussed upon rewarding. Attached a document for essay guidelines. Deadline will be 14 August 2015 6.00 pm (GMT +8) Work will be assessed according to the following criteria which are not evenly weighted, the more important criteria appea...

...reference list. - At least 23 Academic journal, article must be acknowledged by use of accurate Harvard-style in-text referencing; research will be evidenced by inclusion of a reference list. -Compare 2 countries with different industries: Japan’s Toyota (Automobile industry) and China’s Huawei (Telecommunication industry). -Using theories of Dunning’s OLI model, Matthews’ LLL (Triple L) and SIP (Strategic Intent Perspective) theory by Prahalad model to discuss the examples. Details and structure will be further discussed upon rewarding. Attached a document for essay guidelines. Deadline will be 14 August 2015 6.00 pm (GMT +8) Work will be assessed according to the following criteria which are not evenly weighted, the more important criteria appea...

I would like to develop one application which is hybrid combo of skype + what's app + Viber. If you have worked on any chat / SIP / VOIP applications then do get back to me with existing app URLs.

SIP Dialer for Android, IOS, Windows 1. Voice call 2. Message 3. Video call (Front and Back Camera support) 4. Auto speaker/earpiece option 5. Capable of taking keys during call 6. Capable of taking Byte server 7. Option of Proxy server 8. Should be battery efficient and may be closed when not been in use. 9. Should be able to support SIP all codecs and IAX 10. Unique design 11. Phonebook and phone history and messages synchronization 12. Shows balance on the top of the dialer 13. Capable of saving 5 caller IDs and can use any by manual selecting 14. Echo cancellation (with various backends : webRTC, speex, simple) 15. Conference calls up to 4 16. Callback Option (On selecting this option dialer will call on both registered number and destination number) 17....

We seek a professional in the voice over IP industry to set up a phone network for our internal usage, based on 4PSA's VoipNow product. Some tasks for you to perform about this project: - you must choose a phone model convenient for our staff - physical phones to be configured - the selection of a SIP trunking provider - arranging for a voice actor to record our IVR greetings - the transfer of our toll free number over to this new system (currently it is hosted at ) - creation of extensions - setup 4PSA in redundant mode, hosted on two individual servers - access to Linux servers to be used will be provided by us Requirements: - excellent writing skills, as you need to document the setup of the entire system, step by step and detailed, within a wiki page that we will provid...

Using Asterisk: * Analyze pre-recorded voices in an already answered SIP outgoing call to determine state of the call (busy signal, ringback, talking, etc). Main feature is to detect if a call is answered or not. Based on different states, generate AMI events * Build simple control system to manage PIN cards that are to be used the call-flow

I need help for the following scenario, Mobicent Sip Servlet(mss-3.0.564-jboss-as-7.2.0) is installed in and I want to run the example application media-jsr309-servlet. Mobicent is installed in 192.168.0.124. What modification I need to run that application with those two different IP addresses.

The application should have a button to press to send the DTMF digit one, at this time transmits the audio as loose must send or DTMF digit and receiving, must have a contact list. should work with SIP ver 2, compatible with elastix or asterisk, you must link an external Bluetooth accessory, contact names should be linked to sip extensions A similar need for PC console where it can connect to an extension sip

The application should have a button to press to send the DTMF digit one, at this time transmits the audio as loose must send or DTMF digit and receiving, must have a contact list. should work with SIP ver 2, compatible with elastix or asterisk, you must link an external Bluetooth accessory, contact names should be linked to sip extensions A similar need for PC console where it can connect to an extension sip

...Please visit or we are looking similar product. OS: Symbian, Windows Mobile 1) User friendly interface 2) Mobile Dialer should support SIP/H323 for signalling. 3) Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded 5) This software will run on the phone having Symbian or Windows Mobile 5 and 6 having internet via GPRS/EDGE/Wi-Fi/ 3G ... 6) Mobile Dialer should be fully customizable. 7) Able to use Mobile Dialer with any kind of Softswitch,Voip Billing Platform which support SIP/H323. 8) Able to voice / chat with Gtalk, Yahoo Messenger, Skype, MSN, and popular social networking. Please Do not ask for any prepayment Until Project is been tested

...reference list. - At least 23 Academic journal, article must be acknowledged by use of accurate Harvard-style in-text referencing; research will be evidenced by inclusion of a reference list. -Compare 2 countries with different industries: Japan’s Toyota (Automobile industry) and China’s Huawei (Telecommunication industry). -Using theories of Dunning’s OLI model, Matthews’ LLL (Triple L) and SIP (Strategic Intent Perspective) theory by Prahalad model to discuss the examples. Details and structure will be further discussed upon rewarding. Attached a document for essay guidelines. Deadline will be 14 August 2015 6.00 pm (GMT +8) Work will be assessed according to the following criteria which are not evenly weighted, the more important criteria appea...

Hello, we are looking for someone expert in voip , asterisk, a2billing and freepbx to resolve this issue:: 1- in our internal messaging service sometime messages do not be deliver 2- in sip internal calls we dont have a sound

We need an experienced person who can develop a SIP dialer works with Voipswitch and integrate it into Telegram to build a custom version. For this project we need some important features: SIP dialer SIP dialer should be integrated with class 5 softswitch from Voipswitch (please visit its website ) Vippie is a good reference for test an application works with voipswitch. Making and receiving SIP call it should be possible to use liblinphone or any kind of open source library like PJSIP. It should be have best performance on unstable connection like GPRS, EDGE or 3G Loudspeaker. Time, call status logs, last calls log and redial. Support OPUS codec (it should be possible have alternative codec like G729). Proxy is not required, it’s better to...

I look for a provider able to remotely configure a VPN connection on a router Linksys WRT54GL (distribution DD WRT). The ojectif is to bypass a blockage telephonic communications for SIP work: 1. provide a VPN subscription 2. configure the router 3. line validity test SIP tépéphonique

I can not make the configured Elastix (3.0) MT (Asterisk + kamailio) work correctly to make and receive calls with external SIP trunks, they register, but not finalize authentication and stay with status UNREACHABLE. We need a professional who knows the asterisk and kamailio to solve problems encountered with the trunk configuration;

My project focus on Improve th QoS for VoIP in MANET by using SIP protocol as a routing protocol with algorithm to find the shortest path based on R-factor and increase No. of calls at the same time consider the QoS

I need a mobile native dialer app(not a web app). App integrates to all desired phone sections namely; Contacts, call accounts, etc networks...wifi, 3G, 4G, GSM, and employs all call codecs of the phone. 3 screens maximum.......registration, VoIP Dial pad,and In-call visual screen User downloads, automatically installs and sip registers the phone and user can then place a call to my VoIP ITSP server. App should on back-end pull from url the username, password,ip and balance from a database. Thus, user will not have to manually register with those values; our script to app will auto do that for user! Notes: 1. Only experience tech person should come for interview. 2. Time to production is <3 days 3. Freelancing teams/groups not invited. INDIVIDUALS only wanted

I have a 3CX phone system that I need assistance with on configuring with my SIP provider and the softphone

Have a linux server on Amazon Web Cloud, need: 1. asterisk installed and configured 2. install and configure webmin 3. connect to sip termination provider 4. setup first couple extensions and voicemail - voicemail forwarded to email address 5. configure a browser based soft phone (please suggest one or use )

project description Android app should have following feature Features - Goal Planner. - Retirement Planner. - Insurance Needed - SIP Tools- SIP Calculator, SIP Planner, SWP Calculator, STP Calculator. - Loan Tools - Loan Calculator, Loan Refinance and Loan Compare - Fixed Deposit Calculator - Recurring Deposit Calculator - Save Goals and Retirement Plan - View My Plan

We seek a professional in the voice over IP industry to set up a phone network for our internal usage, based on 4PSA's VoipNow product. Some tasks for you to perform about this project: - you must choose a phone model convenient for our staff - physical phones to be configured - the selection of a SIP trunking provider - arranging for a voice actor to record our IVR greetings - the transfer of our toll free number over to this new system (currently it is hosted at ) - creation of extensions - setup 4PSA in redundant mode, hosted on two individual servers - access to Linux servers to be used will be provided by us Requirements: - excellent writing skills, as you need to document the setup of the entire system, step by step and detailed, within a wiki page that we will provid...

Need help with setting up outgoing routs with sip and fxo

we want create small voip dialer hope it be in php or java +will connect to voip accounts +make call automatic +send some digits from dialpad after the call is connected there's alot of opensource code to do that feel free to use any

we want create small voip dialer hope it be in php or java +will connect to voip accounts +make call automatic +send some digits from dialpad after the call is connected there's alot of opensource code to do that feel free to use any

We are requesting a professional who can configure our Nagios and create a template for Network devices and servers. This templates should be used with any devices when i need to add one. It should monitor: - AD DC Service - POP3 Service - SMTP Service - DNS Service - HTTP Service - PING to WAN Router Service - SIP Trunk Status - Bandwidth Comsumtion on WAN of DG - CPU - MEMORY - DISK We request to configure in the same server and to set default aler parameeters We also request manuals on how to add new devices to Nagios.

i want make custom voip dailer from an opensource dailer if you can work on such project please pm me

Hello I just freshly installed Vtiger CRM. i need a simple to install SIP to make outbound voip calls using asterisk (could be another cheap solution if you have any idea). I need someone who has experience with this only and who will be able to explain me how he did so that i can replicate on other vtiger crm hostings i own. thanks for bidding

Need the following done. I have a FreePBX set up already. Need a web GUI where agents from queue can login and off. Web page must be on a seperate server [not on PBX]. Webpage must be password protected with credentials other then SIP. Need the ability of viewing whos currently logged on to the queue. You must provide the code, along with basic instructions on how to setup.

Need the following done. I have a FreePBX set up already. Need a web GUI where agents from queue can login and off. Web page must be on a seperate server [not on PBX]. Webpage must be password protected with credentials other then SIP. Need the ability of viewing whos currently logged on to the queue. You must provide the code, along with basic instructions on how to setup.

# Dialer with unique operator code for run. # Dialer have to be control by cms portal. ex- new dialer create using by new operator code, call registration limit etc. # For VoIP calling, this VoIP dialer integrates with the phonebook of the mobile phone user. # Clients can check their ac...internet connectivity. # Blockage can't stop mobile VoIP, this dialer works efficiently as it can pass through any firewall or blocked network. # Runs behind NAT or private IP. # Client does not need to configure anything except inserting PIN of their account. # Available on multiple platforms: This SIP dialer runs on iOS and Android platforms. # Protocol: Session Initiation Protocol (SIP). # Supported Codecs: g729, AMR, GSM and g711. # Mobile SIP app must compatible with...

# Dialer with unique operator code for run. # Dialer have to be control by cms portal. ex- new dialer create using by new operator code, call registration limit etc. # For VoIP calling, this VoIP dialer integrates with the phonebook o...internet connectivity. # Blockage can't stop mobile VoIP, this dialer works efficiently as it can pass through any firewall or blocked network. # Runs behind NAT or private IP. # Client does not need to configure anything except inserting PIN of their account. # Available on multiple platforms: This SIP dialer runs on iOS and Android platforms. # Protocol: Session Initiation Protocol (SIP). # Supported Codecs: g729, AMR, GSM and g711. # Mobile SIP app must compatible with all standard SIP softswitch. Please call - 0...

I have already...running my VOIP system and trunks. I have cisco SIP phones connected to the Elastix box and an FXO card for my PSTN lines. I use office 365 for my internal communication and emails. I need someone who is passionate about Elastix/Asterisk and knows how to integrate the different tools i mentioned above. so this project is about improving my current VOIP setup to allow the use of all the functionalities. my priority is to fix / implement: - caller ID not always showing on the trunk ( FXO card) - Fax integration with my email server - a shared address book ( LDAP?) that can be used by cisco phones or any soft SIP application connected to the box - integrate my office 365 lync ( now it is skype) with my elastix trunks - setup my VOIP sip provider i...

Hello, I need customize a opensource SIP Softphone. Follow the examples softphones you can use to customize: I need only a function customize (static IP for example) Logo customize Template (colors, skins, etc) - Optionally Mobile Softphone - Optionally

Need a SIP dialer integrate with voipswitch (as same as vippie but doesn't need have other functionality like IM or send file and etc.) Main functions of the app will include: calls through mobile data connection (GPRS, EDGE and 3G) and also WiFi. 2. Integrated contacts from the phone. 3. synchronize phone contacts to server. 4. Time, call status logs, last calls log and Redial 5. STUN 6. Support for codecs OPUS, g729, GSM, g711 and iLBC. 7. Built-in tunnel for passing through voip blockades 8. Enhanced battery usage (should not kill the battery) a. The source code should fully documented. b. Doesn't need develop good UI. Simple UI meets our requirements.

Unistim is already working but one way audio, need to be fixed. SIP trunk need to be configured, its not working for outbound or inbound Fail2ban and iptables need to be installed and configured to block brute force attacks