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    9,716 trunk sip jobs found, pricing in USD

    A mobile dialer app for modern mobile interfaces that should work with any sip voip server using sip protocol and support all the standard codecs, consuming low bandwidth, voip dialing using wifi, 3G/4G and have the functionality to use local minutes (as calling card). user friendly and quality voip calls. User may registered through his verified mobile

    $701 (Avg Bid)
    $701 Avg Bid
    18 bids

    ...webpage, and the colour of most text. List will be provided as a spreadsheet (Excel or Google Docs), to be filled in with Hex Colour codes, captured using an application like Sip. We will provide several examples where we have captured the colours already, and will check in and verify work after the first 5-10 to make sure that you are getting the

    $226 (Avg Bid)
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    19 bids

    I need a sip-phone (IOS, Android, Windows) able to register into Asterisk and peer a SIM in a GSM gateway. Sip Client should be able to send/receive voice calls, SMSs and USSDs. Sip Client should be able to top-up the peered SIM.

    $1005 (Avg Bid)
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    9 bids

    We have conversations (see githup) as basic and inside this we need some modifications. We run our own xmpp server and separates sip server with csipsimple. By api the registration in conversation is there one sip and xmpp users. Under the conversations user can call other user in his contact list. For this there is already some working examples.

    $1350 (Avg Bid)
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    17 bids

    i need an artist to teach a class at a paint and sip I'm hosting on Feb 10th for 2 hours

    $107 (Avg Bid)
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    7 bids

    I will give access to a linux box, you can install asterisk along with any other services, I want a web interface, where i will copy paste 200 phone numbers, also there should be option to upload a voice record (audio) , once i click play button. Each of the 200 numbers should get a call from their own numbers (spoof) simultaneously and all of them should hear the audio file played. Please ...

    $504 (Avg Bid)
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    3 bids

    Set up an auto Fail-over trunk on asterisk, trunk will be given. Looking only for expert in asterisk to work with.

    $105 (Avg Bid)
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    10 bids

    Skill Pre-Requisites: Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory)

    $19 / hr (Avg Bid)
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    4 bids

    We have our own sip / xmpp server and we have compiled the latest build of csipsimple and Linphone. We want a secured voice over zrtp. This is a peer to peer encryption but i dont get it work, not under csipsimple and not under Linphone. I need support to get this work. Linphone and csipsimple can be downloaded on playstore.

    $157 (Avg Bid)
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    1 bids

    hi, i need to configure trunk sip for outbound campaign and inbound call. im stuck in dialplan setting. in asterisk debug i can show all circuits are busy. i wait your reply tanks

    $19 / hr (Avg Bid)
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    13 bids

    Call functions like mute, conference, hold, transfer, and call rec...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 SIP Webphone

    $504 (Avg Bid)
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    15 bids

    Dear all, I used to have Elastix with CISCO 2800 as FXO gateway, now I have issabel instead of Elastix and moved all the trunk parameters along with the outbound route yet calls are failed to connect , seeking professional help thanks

    $25 (Avg Bid)
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    3 bids

    i simply want 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

    $1272 (Avg Bid)
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    19 bids

    ...website comprising most of the items in this diagram attached: 'schematic minus BTS [url removed, login to view]' . The diagram should comprise a "MultiDSLA System; DUT UE; IMS; VPP+ (Reference SIP Client); DSLA (Analogue test interface); a cloud". similar to '[url removed, login to view]'. diagram should be in a style similar to the 3 examples given be...

    $60 (Avg Bid)
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    Trophy icon Architecture Design 3 days left

    ...to each other branch Two branches have slides down to pool in middle of tree or center of tree This Resort must be 30 Floors tall. With 10 to 20 rooms on each floor so trunk of tree resort is MASSIVE It will be brown with only few leaves. 3d or digital is fine as long as I can see outside from different angles like from front and back I want

    $500 (Avg Bid)
    Featured Guaranteed Sealed
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    7 entries

    Dear all, I used to have Elastix with CISCO 2800 as FXO gateway, now I have issabel instead of Elastix and moved all the trunk parameters along with the outbound route yet calls are failed to connect , seeking professional help thanks

    $26 (Avg Bid)
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    3 bids

    I need you to develop some software for me. I would like this software to be developed for Windows using .NET. Sample C# Project to :1. Connect to SIP trunk2. Make a call : play a file when pickup , hangup after playing , run event when hangup(by system or user) or no answer3. Receive call : play a file , run event when hang-up or no answer (by system

    $1049 (Avg Bid)
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    ...[url removed, login to view] [url removed, login to view] to style and layout on the first login page of wifi visitors ([url removed, login to view]) [url removed, login to view] a social login option via instagram and linkedin. [url removed, login to view] additional social login information to the [url removed, login to view] only email and name are saved. Please see

    $138 (Avg Bid)
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    Hi, I'm trying to setup a Debian Jessie SIP server with Kamailio. Everything works but the TLS handshake doesn't complete. It stops at the client handshake, so the server doesn't send it's certificate. I would like someone who has experience setting up Kamailio with TLS and unix server administration. The deliverable would be to let me know

    $116 (Avg Bid)
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    3 bids

    Must be experience in Open Source IP-PBX development product. Setting up the infrastructure, build key features and testing. Works on Debian® 2.8 distribution, some of the key FOSS components that support the unified communication and management functionality are Asterisk®, FreePBX®, Chan_Dongle, and RaspAP Web GUI. Build a IP-PBX product that allows users to make VoIP calls to...

    $1226 (Avg Bid)
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    18 bids