Softfone g729 työt
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). 2) Send and recieve calls to saved numbers automatically for adjustable time. 3) stop working if calls not con...
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). 2) Send and recieve calls to saved numbers automatically for adjustable time. 3) stop working if calls not con...
Mobile VoIP Dialer for IPPBX Features required: Call to extension or call to a number or contacts Integration with Mobile Phonebook Call Transfer Call Hold Call conference ./ multiparty conference call G729, GSM, iLBC, G711, G722, AMR Codec support Tunneling will be considered as additional value
Mobile VoIP Dialer for IPPBX Features required: Call to extension or call to a number or contacts Integration with Mobile Phonebook Call Transfer Call Hold Call conference ./ multiparty conference call G729, GSM, iLBC, G711, G722, AMR Codec support Tunneling will be considered as additional value
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
We are looking for solution like a traditional GSM or CDMA VoiP gat...and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...
...convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in ...
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK form...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK form...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...
Project Description I need an application that runs on Android phones and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. If android Layers have a problem for this job, you can use replicans OS. The applic...calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...
...WHAT IT IS ABOUT send query before bidding Server X = Asterisk Server Server Y = Window based Asterisk Client server Main goal minimize Bandwidth in client side with quality voice . Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity
...tunnel between two asterisk servers to compress and bypass voice packets, we will provide some demonstration of the existing services to the candidates after interviewing and once we believed that you can do it. Server A = Asterisk Server Server B = Asterisk Client Explanation of Scenario: 1. Server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted...
I want to add codec G729 to increase quality call on my system and make a modification on my server, check my setting to ensure i got them right. Currently cannot make international calls, suspect codec as my providers use G729, G711, G711u etc and not ulaw or alwaw which are default for a2billing. A few small bits here and there.
...parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on wifi Internet connectivity and mobile Data internet , All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance through USSD all cell phones will act as a port like any normal vo...
...parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on wifi Internet connectivity and mobile Data internet , All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance through USSD all cell phones will act as a port like any normal vo...
We are looking for solution like a traditional GSM or CDMA VoiP gat...another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...
A Sip Dialer App is required with g729 codec for iPhone and Android. App should have functionality to change settings for codec, jitter, communication ip/port and add/remove account setting for customer.
im looking for a soft phone application for desktop, android -auto provision -contact -good quality sound -low bandwitdth G729 H264 -TLS / SSL -all business features ( trans, conference,hold, mute........ ) VOIP- ANTIBLOCK IP AND TRAFFIC COMPRESSION solutions for censored countries such as egypt, Iran, China call center soft ware with all the call center features FOP2 trans, recording,trans, blind trans, etc..........
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
I need an application that runs on Android phones (4.0 or above) and can 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 conv...calls from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. Sip Requirements - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all calls made Skills required: And...
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
Desenvolver softfone para os sistemas Android, iOS e Windows PC com funcionamento semelhante ao ZOIPER Softphone (Android e iOS) e Microsip (Windows PC). Este softphone deve ser capaz de registrar contas VoIP e aceitar as configurações de Criptografia de um Servidor (TLS e SRTP).
...GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels if they are busy. - If a channel is busy the call shall try automatically the next channel and so on till it finds a free channel. - GUI interface to monitor and configure the system -sim card sus...
Hello, need someone expert to install C5 Switch and doing this parts: Enable Voice and video calling Enable G729 Codec And add SIP Trunk
Expert in C, C++, PJSIP Stack Must be able to analyze the Wireshark captures to identify any SIP signaling/media issues. Should have worked on Server side Should be aware of RFC 3261,3264, NAT traversal, Media Codecs AMR, G729, Opus.
Hello guys, This project is belong to who make are expert. i need to install FreeSwitch on my server and install ASTTP on that. then doing this configs: 1- Activate G729 Codec on that 2- Change transfer protocol to UDP 3- activate just SIP protocol on server. (disable another protocols) 4- Enable Video Call feature on freeswich if you not robot please add "Swh" in your bid. Thanks for your attention
REQUESTS: 1- compile G729 codec for asterisk 1.8 on mips device 2- compile new version of FFMPEG because old version not work with online link example: root@arc-pbx:~# ffmpeg -i ...... libavutil version: 49.6.0 libavcodec version: 51.55.0 libavformat version: 52.13.0 libavdevice version: 52.0.0 built on Jun 18 2012 18:38:26, gcc: 4.1.2 : no such file or directory Device : MIKROTIK Metarouter mips other details: KAMIKAZE (8.09.2, r18961) * 10 oz Vodka Shake well with ice and strain * 10 oz Triple sec mixture into 10 shot glasses. * 10 oz lime juice Salute! --------------------------------------------------- root@arc-pbx:~# uname -a Linux arc-pbx 2.6.31.10 #6 Sat
Need Linphone Customization for Iphone. requested features as bellow. 1.) add G729 2.) customize the interface (we will provide the required images for it) 3.) need to add an in-aPP signup and registration with users phone number. All the required web services will be provided. can also provide an old version of the completed software 4.) users can re login using the phone number and vitrifying an SMS. Note : all the UIs and images will be provided, on successful completion it must be uploaded to APP store. any changes required in the uploaded process has to be done.
We are looking for solution like a traditional GSM or CDMA VoiP gat...another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...
I want someone who can install and configure goautodial 3.3 with free codec G729 for my new call center
We are looking for solution like a traditional GSM or CDMA VoiP gat...another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity, coz GSM internet data normally disable during any GSM call. All call will pass with G729 codec The registration server may be Asterisk or VOIP switch or any other server or customize server. This server will receive call from another VOIP switch server with SIP protocol. Certain number of registered Mobile will be able to assign in a group of gateway. On server have to include option to show balance th...
...using RADIUS protocol. It should send a RADIUS message to our billing server when a call arrives, when outbound is connected, disconnected, or gets a proceeding/progressing/alerting message, when a digit is detected on either leg, or when either end hangs up. Route outbound calls through a series of outbound carriers, as instructed by billing server Support codecs: • G711 u-law • G711 a-law • G729 • G729b • G723 • G723a • G726 • GSM • iLBC Web interface should show current activity and CDR, which should be searchable by date/time, originating IP, destination number, destination country, or outbound carrier. For any call in the last 4 days, display full SIP logs by clicking the line in the CDR, plus a direct link to the...
Need Linphone Customization for Iphone. requested features as bellow. 1.) add G729 2.) customize the interface (we will provide the required images for it) 3.) need to add an in-aPP signup and registration with users phone number. All the required web services will be provided. can also provide an old version of the completed software 4.) users can relogin using the phone number and vitrifying an SMS. Note : all the UIs and images will be provided, on successful completion it must be uploaded to APP store. any changes required in the uploaded process has to be done.
Preciso alterar uma tela do módulo de call center do Elastix 4.0 . Preciso que na tela de monitoria de agente apareça o número do telefone que o agente está falando e preciso colocar um ícone de um alto falante nessa tela para que ao clicar nele o supervisor possa ouvir a conversa sem a necessidade de digitar 555+ramal no softfone. Aguardo retorno.
Hi MikeRRR, I have a problem with a new install i made of asterisk (freepbx), and i am not being able to install g729. Also i think i have some problem with the codec translations (as core show translations give a strange result). Are you able to help?
Hi Asterisker, i'm having a problem with a new asterisk server we installed, and not being able to install g729 codec. I am also receiving error on the translations of codecs. Are you able to help?
The following is require to complete VoIP Server Project 1- Build Asterisk with A2billing 2- Customer Frontend design to allow landing user Page layout for custoemrs to be user friending to register and make phone calls. 3- Setting up VoIP Trunks With VoXBEAM Making sure 4- G729 Codex will be required. 5- Registration / Emails / Payment method setup 6- Source code of any CSS or web dev shall be retained and called from the server . Please do not bid if you don't have previous record or experience. You will have to share demo. IF YOU BID YOU NEED TO BE SURE YOU HAVE READ ALL PROJECT REQUIREMENTS BEFORE PLACING YOUR BID Thank you
Project Description: 1. Brand Icon and Company Name on the application 2. Add account Basic option will be used .Username - Enter your Username Server - our server address and port will be hardcoded [so Server option will ...1. Brand Icon and Company Name on the application 2. Add account Basic option will be used .Username - Enter your Username Server - our server address and port will be hardcoded [so Server option will not be anymore available here] Password - Enter your password 3. Call logs will be only numbers - not the whole sip URI 4. Under Settings -> Media -> Codecs -> G711a,G711u, G729 will be enabled by default 5. Help : Link to our web site 6. when call in progress - showing SIP/Account name : phone number [ SIP/ te...
The following is require to complete VoIP Server Project 1- Build Asterisk with A2billing 2- Frontend design to allow Page layout customers to register and make phone calls. 3- Setting up VoIP Trunks With VoXBEAM Making sure 4- G729 Codex will be required. 5- Registration / Emails / Payment method setup 6- Source code of any CSS or web dev shall be retained and called from the server . Please do not bid if you don't have previous record or experience. You will have to share demo. Thank you
The following is require to complete VoIP Server Project 1- Build Asterisk with A2billing 2- Frontend design to allow Page layout customers to register and make phone calls. 3- Setting up VoIP Trunks With VoXBEAM Making sure 4- G729 Codex will be required. 5- Registration / Emails / Payment method setup 6- Source code of any CSS or web dev shall be retained and called from the server . Please do not bid if you don't have previous record or experience. You will have to share demo. Thank you
Hello We need Simple Encrypted App for Android. Requirements: Logo DTLS-SRTP Support green padlock that confirms that the call is encrypted G729 Please Bid if you can give demo of the App
Hi kingAsterisk, I want to configure my cloud server A2billing with freepbx having code G729 and TLS can you please code me your best price I can give u server on centos 7
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK form...GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or inband - Use codec G711 and GSM - Be able to use codec G729 - Receive through GSM one 1 simultaneous call and rout it to a predefined SIP client - Be able to store a CSV with all ...
...termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Other standards codecs like G722, G726, alow, ulow... shall be supported - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels if they are busy. - If a channel is busy the call shall try automatically the next channel and so on till it finds a free channel. - Fre...
Hi Guys I want you to make a sip Dialer capable of making and receiving voip calls with G729 Codecs and branding. All udp data packets should be sent in an encrypted format I will provide the encryption code in c/c++ , similarly all incoming udp packets should be decrypted. This encryption and decryption should be applicable to rtp stream also. If you have any existing dialer and can add encryption decryption function provided by me in your dialer that will also work. I will need full source code on completion along with the documentation. I will prefer someone who has done similar projects. Thanks Vidhya
...to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.... This will be main future.. and it will work on codecs g711 g729 g723 I KNOW ITS NOT POSSIBLE IN ANDROID BUT MANY PEOPLES CONFIRMED THAT ITS POSSIBLE IN ANDROID WITH CHIPSET MTK6577. ____________________________________________________________________________________________________________________________________________ Other then this i want you to add some human behavior options.. 1) pause between calls (adjustable in settings). ...
