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    16,382 cisco sip config jobs found, pricing in USD

    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)

    $20 / hr (Avg Bid)
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    Looking for Freelance CISCO expert to review the current IVR flow, suggest optimization and best utilization as a call center solution with IVR self-help.

    $650 (Avg Bid)
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    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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    VOIP 6 days left

    Cisco Ip phone configuration,implementation,imprensence,expressway c and e,unity server.

    $1045 (Avg Bid)
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    I am currently developing a training class for Cisco Firepower. I need support in creating powerpoint slide decks for the different lab modules I am developing. Based on planning of the class I need slide decks for roughly 14-15 hours of teaching. This will result in 800 to 1200 slides, depending on how much content is put on each slide. The general

    $1718 (Avg Bid)
    Featured
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    Buidling a HP Cabinet with 3 C700...to handle the traffice of the management cards (VLan 20) and the other to handle the traffic from the blades to the internet(Vlan 100) .In each C7000 chasis there is a HP Cisco 3120 switch with 8 ports. The traffice from the 16 blades will need to be mapped to the 8 ports and then they can be connected to Vlan100.

    $32 / hr (Avg Bid)
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    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

    $20 / hr (Avg Bid)
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    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

    $522 (Avg Bid)
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    ...use the config for the other sites. the config holds lots of info like GA, db, facebook, site links, names, etc My thought process and what ive done it mad a folder called /cfgs/ and put some files for towns in there *[url removed, login to view] and [url removed, login to view] so basically we need the site to be directed to use the different config files if

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    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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    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

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    ...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...

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    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

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    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

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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

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    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...

    $1145 (Avg Bid)
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    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

    $20 / hr (Avg Bid)
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    1 bids

    It's simple I will rent a Linux Server (Centos u Ubuntu) and I n...(Centos u Ubuntu) and I need to install a dialler in it. The dialler is an Asterisk based one and all it's suppose to do is send call automatically to a telecom server by using a SIP account with 30 concurrent calls capabilities. The account has already been set in the Telecom Server

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    15 bids
    ASTPP Fix errors 4 days left
    VERIFIED

    ...I am new on ASTPP. I use the link([url removed, login to view]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE: IM002

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    Ubuntu VPS running on Apache2 and Virtual Host needs the following config and setup: 1. Configure Apache2 config file to run SSL and tighten security 2. Configure LetsEncrypt 3. Check 2 Vhost configs, and enable SSL from Letsencrypt (and create a working default for later copies). Enable Vhost to open from different IP adresses (each site has its

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