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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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    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

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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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    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

    I need to install and configure SMTP SSL / TLS on Ubuntu VPS. Configure records A, MX, DNS, DKIM, SPF, DMARC, PTR for domain. After the installation I will need a script that will do a lot of this setup automatically. A readme file will be needed also. When you bid, please provide me with this code( vpsSMTPinstall ) in your bid message , or

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    Urgent
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    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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    ...webservice that will do the following 1. Receive a soap request, log it sql data base, convert it to a fixed length string 2. Push the message to a remote IBM MQ server SSLV3 with TLS 1.2 3. Receive response, convert fixed string response to SOAP message, log to SQL 4. String to Soap will be defined on a XML configuration file 5. log messages to SQL database

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    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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    ...server enabled to respond to network pings • RTCC is configured and is updated using SNTP client against [url removed, login to view] • Has minimum TLS v1.2 to port 8883 (port is adjustable within a header file) • Can do non TLS on port 1883 if required • Uses the cryto engine of the microprocessor via Harmony framework if achievable • Where possible,...

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    ASTPP Configuration 3 days left
    VERIFIED

    ...create origination rate table and termination rate table 4. create trunk and set up routing so that origination carrier calls cam be terminated to termination end. 5. create a sip user account on new customer and route his calls to trunk of termination carrier. test and make sure all calls connect properly. give me a walk thru of the steps taken to

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    ...solución, implantación y soporte. 3. Conocimientos técnicos requeridos: a. Redes IP: routing (BGP, IS-IS, OSPF…) b. Seguridad: DDoS, seguridad en protocolos de red (HTTPS, SSL, TLS) c. Fabricantes: Arbor, Radware d. Idiomas: inglés (experto) 4. Conocimientos deseables a. Redes avanzadas: MPLS, MP-BGP, VPN, GRE b. Seguridad: a nivel de aplicaci&o...

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    ...Hosted PBX and Call Center. Based on Asterisk/Freepbx /Freeswitch etc. I would like to set it up in Amazon AWS so that we do not have to worry about the servers. Will have SIP trunking for the incoming and outgoing calls. Each customer will have their own portal to manage their users, extensions etc which can be done either with freepbx or if you

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    Buenos días, Buscando una persona para crear varios manuales/guias para la configuración de una cuenta SIP en centralitas Asterix y otras del mercado. Necesitamos una guia de configuración basica y con posibles errores - soluciones. Gracias

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