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    465 iax software visual basic travaux trouvés au tarif de EUR

    Bonjour, je chercher un développeur pour coder une application sous android et IOS avec les spécifications suivantes : 1- Appelle sortant via un server(SIP/IAX). 2- Forward les appelles Voip vers GSM avec un limitation de duré max 10 minutes/24 heures( Call voip ---->GSM) 3-l'application doit être exécuter sur le smartphone en arriè...

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    Write an Android application S'est terminé left

    IAX /SIP Dialer , codec support G729,GSM,G711

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    ...updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and setup

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    ...updates all other currencies to cost based on the ZAR price through the standard Yahoo Finance Integration already present. 2. Ability to create new Realtime ChanSip, PJSip and IAX accounts that FREE PBX can connect to without needing to re-load A2Billing. (DB Accounts) This should however not affect the Free PBX configuration 3. Configure and setup

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    VIBER WHATSAPP SIP GATEWAY S'est terminé left

    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber/whatsup protocol 3) the termination number carried from the origination header will be checked by the Viber/whatsup gateway , if the number is used by Viber/whatsapp

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    Project for DataHome -- 2 S'est terminé left

    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    LDN
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    Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access

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    Project for Anton F. S'est terminé left

    ...you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry via USB bluetooth dongles. We have new installations or needs of support for occurring problems. We

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    ...calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

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    ...to send missed calls (outbound calls to be disconnected after 1-9 second of rings) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the following requirements: 1. We are looking forward to send massive amount of missed calls using hundreds and

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    Need a simple design for [se connecter pour voir l'URL] and it can literally be ju...simple design for [se connecter pour voir l'URL] and it can literally be just a font treatment and then a wraparound box design. Similar to [se connecter pour voir l'URL]

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    ...cliente VoIP para sistema operativo Android que posea las siguientes características: 1) Para versiones android 3.2 en adelante 2) Que permita conexiones mediante protocolo IAX (Asterisk) y SIP 3) El cliente se utilizará sólo para contestar llamadas (no debe contar con marcador) 4) El APK debe poder actualizarse remotamente por nuevas versiones, ya

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    Viber / whatsapp sip gateway S'est terminé left

    ...one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if the

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    ...to send missed calls (outbound calls to be disconnected after 1-9 second of rings) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the following requirements: 1. We are looking forward to send massive amount of missed calls using hundreds and

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    ...Presence/online status Call history Voicemail messages and missed calls notifications Contact integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 RAW SIP technology, not

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    VIBER WHATSAPP SIP GATEWAY S'est terminé left

    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    ...Presence/online status Call history Voicemail messages and missed calls notifications Contact integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper

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    This is what the company makes. Sells for kids through their parents (who actually make the purchase). Imaging should give the feeling of fun and safe. ...give the feeling of fun and safe. Need full color and black and white. Colors defined, size and ratios. Vector art. This is slime: [se connecter pour voir l'URL]

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    ...-implementation) 3. Number of Server Y can be unlimited. 4. Number of Gateways/E1 cards per server Y can be unlimited 5. For server Y installation will be window based software based on Asterisk can be easily downloaded from our website and can be installed must be in [se connecter pour voir l'URL] file mode ([se connecter pour voir l'URL] for reference). This installation

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    Project Description: We are looking forward to develop an application that is currently being served by few companies (see the list below) as far we have studied, this software do 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

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    Telefonia Issabel S'est terminé left

    Se busca persona para configurar un servidor Issabel (elastix). Alrededor de 100 internos. Cada interno deberá tener una extension sip y una Iax (ej interno 100 sip y 500 iax) Cada interno deberá poder administrar desde la web su extension, deribacion de llamadas, pero no ver los menu de los demas (configurar los permisos OK). Se deberá colocar

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    Trophy icon Eden Hope Academy "Dollars" S'est terminé left

    Use our logo to design "dollars" for classroom use. These are like Monopoly money, [se connecter pour voir l'URL] 8.5 x 11 Black and white only 10 per page $1 and $5 Must include: Eden Hope Academy

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    Trophy icon Logo for Wholistic Women S'est terminé left

    ...subjects) as well as for booking client appointments in person (mobile) and potentially online. I love the colour cherry red e.g. [se connecter pour voir l'URL] but realise sometimes that is not always the best for advertising

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    WHATSAPP VIBER SIP GATEWAY S'est terminé left

    ... triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    VIBER WHATSAPP SIP GATEWAY S'est terminé left

    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    ...for VoIP, SIP, IAX who will explain us the feasibility of the following requirements. It is therefore only a question of describing whether or not and if so how, the following is possible: We want to develop a native mobile app for iOS and Android. This app should be able to make a VoIP call (please recommend open source VoIP, SIP, IAX code to use)

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    I have a new Asterisk installation and I need to configure incoming calls for IAX and SIP.

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    ...friendly interface (Design Guides to be supplied by us) 2) Mobile Dialer should support SIP -V2 & IAX 3) Mobile Dialer supports G729, G722, GSM, G.711,OPUS 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling (embedded ?) 5) This software will run on the phone having Android and IOS Must also operate over the following mediums GPRS

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    ...calls. Functional flow 1) Calls originating will send to Viber, Whatsapp, WeChat and Facetime gateway 2) Viber, Whatsapp, WeChat and Facetime gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber, Whatsapp, WeChat

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    À la une Urgent
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    Trophy icon Design a Logo for a android app S'est terminé left

    ...it can be colorful or black and white. This software is an android app. Name is Clerk. What is a Clerk? check this [se connecter pour voir l'URL] You need to take in account the definition and image references in the link below, as ideas and guideline. [se connecter pour voir l'URL] [se connecter pour voir l'URL]

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    Project for tektrix1 S'est terminé left

    ...one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    complex asterisk setup S'est terminé left

    i need to setup asterisk server to rec...different prefixes and then the switch send the call back again to astersik which will forward it tol client through iax trun on the client side there is a pc with centos installation before the gateway on this pc we will install iax and the pc will forward the call to the gateway using sip protocol on lan

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    Project for xeeonix S'est terminé left

    ...codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc Functional flow 1) Calls originating will send to Viber gateway 2) Viber/ gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber and if the number

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    SIP to viber, whatsapp, Line S'est terminé left

    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    VIBER WHATSAPP SIP GATEWAY S'est terminé left

    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    Write some Software -- 2 S'est terminé left

    ... we will used . A. iax trunks in trunking mode. B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc. we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can

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    Write some Software S'est terminé left

    ... we will used . A. iax trunks in trunking mode. B. Open vpn static mode and dynamic mode C. Tnic static and dynamic mode 8. Asterisk web billing gui for adding gateways. Adding client , Prefix , dialing plan viewing active calls, billing cdr ,etc. we will provide you the Dedicated server asterisk and client asterisk configure IAX trunking, so we can

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    Hi right now i am running asterisk my server in USA and My local PC in Bangladesh I m connected asterisk to asterisk useing IAX2 NOW per call 8KB on g729 codec BUT the prob is my country ISP blocking my server IP so i need a solution i try to use sip . its working fine but sip use per call 20KB i need under 10KB per call i try to use openvpn but if we connect openvpn its got block in...

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    Trophy icon Design a Logo for a software app S'est terminé left

    ...colorful or black and white. This software is an android app. Name is Excursion. You need to take in account the definition and image references in the link below, as ideas and guideline. [se connecter pour voir l'URL] [se connecter pour voir l'URL] Logo formats: hdpi

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    Building Asterisk Main Server Connecting Main server with existing Two Asterisk Boxes via iax2 trunk mode ** (One of the advantages to using the IAX protocol to do this is a feature called trunking, which utilizes a method of sending the voice data for multiple calls at once with a single header. This has little effect on only one or two

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    ...backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE . Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and

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    À la une
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    ...Server Server ( Main Server ) Client ( Puppy Linux ) Goip ( GSM Gateway ) Server has public ip and client is behind the NAT Both running Asterisks and trunking is IAX Also Client is connected though openvpn with server for Encryption Server getting Voip Traffic from Voip swtich and passing to Client and then client pc send calls to Goip

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    FreePBX Call Accounting S'est terminé left

    ... 1. Setup rates for trunks in FreePBX. This should allow for rating numbers prefixes. The system should match the closest prefix. It should rate all types of FreePBX trunks iax, sip, dahdi and others. It should also rate based on time of day and specific increments. 2. Importing and Exporting of Rates - Rates should be able to be exported to csv

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    asterisk Bandwith problem S'est terminé left

    hello i have a problem on asterisk now i m useing asterisk 11 when i m useing asterisk iax to iax connecting codec g729 its useing per call 8kbps but when i m useing SIP to SIP connecting codec g729 its useing per call 25kbps also i try g729:200 its makeing voice quality low but not much compressed need help if you can fixed it plzz bid

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    asterisk SRTP install S'est terminé left

    Hello i have asterisk server now i want to install SRTP you have to install SRTP on my server and teach me how ...SRTP why i m useing SRTP local isp blocking my ips thats why before i use openvpn if you have any Batter idea You can tell me I want to Install SRTP and connect IAX to IAX | SIP to SIP | sync to SYNC if you can please bid

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    VIBER WHATSAPP SIP GATEWAY S'est terminé left

    ...triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if

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    A2billing on Asterisk Fix S'est terminé left

    ERROR[20100]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("", "(null)", ...): Name or service not known Sip Channel(s) P...alaw Rx:NEW Tx:ACK (None) [se connecter pour voir l'URL] (None) 11718/00118 00002/00001 00000ms -0001ms 0000ms Unknow Rx:REGREQ Tx:LAGRQ 14 active IAX channels

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    Numbers Test DTMF signals S'est terminé left

    We Need To develop one Web base or desktop application with SIP or IAX protocol for phone number testing with Mailbox 1-Uploade Numbers 2- register sip or iax account 3- dail number 1 by 1 from uploded list 4- When call answer send DTMF signal (* & *# ) if after seding this signal system get possitive reply from dailed number save

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    PROJECT DESCRIPTION We want to build an Asterisk bandwidth optimizer for VOIP voice traffic - using IAX. This will use standard software, we assume linux and IAX. (We are open to suggestions.) We want to hire someone who has already done this. If you haven't already done this before don't bother applying please. We want something similar to SBO or RBC

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    ...through a cellphone running android. 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 or IAX to GSM channel/trunk/FXO and other SIP devices can connect and make phone calls using this. You choose the best way to do it: Add the feature to SIPDroid: [se connecter pour voir l'URL]

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