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    10,709 sip client development mobile phones δουλειές που βρέθηκαν, με τιμές EUR
    Fork OpenNMS to support asterisk-java 2.0 and PJSIP 6 μέρες left
    ΕΠΙΒΕΒΑΙΩΜΕΝΟ

    ...com/OpenNMS/opennms) is an open source monitoring and systems management platform written in Java. It includes support to monitor SIP devices registered to and asterisk server and it does it by connecting to Asterisk's AMI interface and querying for SIP registrations. If the list of registrations change, OpenNMS generates an event or an alarm. OpenNMS depends on a

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    Good day, I am looking for a resource to configure queue metrics and freepbx, this is a new clean setup, new boxes. This you will start with clean setup. We have already a sip trunk provider and there accounts need to be setup also Both boxes is already loaded, we only nee to finish off the setup and configuration. We wish to extent both products

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    Free PBX inbound route 6 μέρες left

    If possible, I would like to route an inbound route based on TRUNK and NOT based on DID. FYI: My SIP provoider is NOT sending the DID, so the routing should go based on TRUNK (thus when an incomming call on trunk A comes, then it should be routed to extension 200, when an incomming call is comming on trunk B, it should be routed to extension 201 ect

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    setup an uninstalled NEC SV9100 PBX with two phones and access a single PSTN Australian Telstra Line System has NEC licences from prior owner. PBX HAS has GCD-CP10 GCD-4COTC GCD-8DLCS GCD-4LCF PHONE ARE DT400 and DT800

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    Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.

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    I need some help with my business.

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    Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.

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    We have been developing large variety of “Personalization Apps”. “Launcher App” is a kind of app to be able to customize wallpaper and icons of Android smartphones. This time, we are looking for graphic designers who can make “Launcher themes”.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    AXNET VoIP 2 μέρες left

    ...looking someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones like cisco, Unifi VoIP phone and Polycom

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    Hello, We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough

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    Trophy icon Study Cafe Name is needed!! 4 μέρες left

    NAME IS NEEDED. NOT A LOGO!!! I NEED A NAME SUGGESTION Study Cafe Name needed! - This is a space where you can study and get sip of coffee and some desserts. - space is mainly for studying (and maybe working) - there are space for them to do team work - there are space for customer to rest too - please see some example pictures.

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    VOIP Calling System by using Js SIP Library in ASP.NET MVC 01 1 μέρα left
    ΕΠΙΒΕΒΑΙΩΜΕΝΟ

    I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.

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    We are looking for an artist to create desirable pieces that the general public would like to sign up and...instructions to replicate which we can share with our local artists to replicate in a classroom setting (photo+text). Hoping for quality work very similar to the top paint and sip companies out there, and would be willing to work with long term.

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    Configure OPENSIPS 1 μέρα left

    ...an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5. - The SIP signalling shall transport a 180 RINGING message without SDP parameters meaning the originator should

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    ...API 2. Call is initiated via Twilio Outbound Voice API using a Phone Number. 3. When call is answered, the call is forwarded to another Twilio phone number that uses Elastic SIP Trunk. It is attached to our IBM Voice Agent Service. 4.. Once Voice Agent is connected the user will speak to it., then be transferred to another phone number. We are looking

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    Design on T-shirts, mobile phones and other clothes, and choose the drawing of your choice.

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    Configure a Cisco as5400 with an ss7 link 7 ώρες left
    ΕΠΙΒΕΒΑΙΩΜΕΝΟ

    We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Need Asterisk expert, Phones are down with new router trying to debug where the problem is

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    kamailio5 SIP confing with RTPproxy Έχει λήξει left

    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [κάνε είσοδο για να δ&epsi...

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    Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want t...freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

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    I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

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    ...create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want to make it under 10KB we need to remove or change RTP payload or you can make a SIP client to rerouting call change PTIME or if you want you can do anything else as you want we have only a openwrt router in bangladesh and you can install small package like any SIP CLIENT

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    Hi everyone, I've created an Android application which is linked to my website us..."device_name":"Suzuki Swift | Y 98-4","email":null,"mobile_phone":null,"command":null,"overspeed_speed":"20","overspeed_distance":1} I need this application available for Android phones and IOS phones. I will be lo...

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    I have a bunch of phones that I need to have auto firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they

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    Backend Developer with PHP/SMPP/SIP Έχει λήξει left

    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

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    BSE STAR MF PHP Web Application Έχει λήξει left

    ...[κάνε είσοδο για να δεις το URL] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight)...

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    phones Έχει λήξει left

    I need an Android app. I would like it designed and built.

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    Application key features: * high definition audio and video calls * audio c...sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers allowing to reach everyone that has a "classic" phone line.

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    Homer Seven SIP Capture Έχει λήξει left

    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

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    FreePBX Setup Έχει λήξει left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

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    Asttp Support (Freeswitch) Έχει λήξει left

    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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    SIP RTP PayLoad remove or change Έχει λήξει left

    i need a software to remove or change SIP RTP PayLoad like if i install this software on os it have to remove or change all incoming and outgoing SIP RTP PayLoad if you can make it please let me know

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    Mobile VoIP Dialer Έχει λήξει left

    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    hello guys, i have a curl script that dont work with all mobile devices

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    solución pbx Έχει λήξει left

    Tengo una centralita pbx con issabel y 2 troncales sip con gateway goip en redes difefentes pero siempre se desconectan y tengo que reiniciar el servidos pbx para que se vuelvan a restablecer

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    need hobby geeks Έχει λήξει left

    Looking for an On the Call Technician, who has some ex...6. Basic Computer Knowledge 7. Have a laptop or desktop with windows 7 or higher. 8. Photo ID 9. Paypal account for payments 10. Headset Optional but preferred. for windows sip phone, or android, iphone 11. willing to be creative, and outside the box thinker 12. Honest Contact me to Learn More

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    ...server under elastix and the server have 2 network card One card link to Sip line from local Telecome company and another network card for internet connection My local server is online now via port forward from the router and it’s working by Zoiper app very fine in computer an mobile... in local network and online internet network But the problem its

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    ...running local FreePbx server under elastix and the server link to Sip line from local Telecome company and there is another network card for internet connection My local server is online now via port forward from the router and it’s working with Zoiper app very fine in computer an mobile... But the problem its just working with Zoiper nothing else So

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    I won't disclose too much information about the project but my objective is create a piece of software for internet security, to be used by anyone, I would like it be a subscription service and very modern accesable to all regardless of understanding of computers. I would like the website to be codded from scratch no wics or squarespace but this point is up for discussion if you belive it is ...

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    Mobile Hybrid SIP calling APP Έχει λήξει left

    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or phonegap or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through a SIP proxy. we will provide all the server details needed

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    Description. A simple PBX pure software would have to be [κάνε είσοδο για να δεις το URL] software should be installed on Windows Caller calls the number to Will and can select appropriate department through IVR. The language should be stored under MP3. Customer lists to maintenance...

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    I want to add new SIP domain in skype for business and make it working.

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    VoIP SIP Intercom Module Έχει λήξει left

    ...Mbytes SDRAM 4 Ohm Speaker Driver, and 2 UARTs 2 channel Audio Codec with sampling rates up to 48kHz Industrial Temperature: -40°C to +85°C Software Features Fusion RTOS TFTP Client Telnet Server Voice Engine with RTP (includes support for RoIP products) Voice Engine Codecs: G.711 (µ-law, A-law), G.722, G.726 (16/24/33/40 kbps), G.729, DVI4 (narrow/HD/Ultra

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    Art History Essay Έχει λήξει left

    ...Today our fans want to ask you a few questions about some of your paintings and the media you have used. Daumier: Well of course. Daumier picks up his water and takes a small sip. Presenter: The first painting our viewers are interested in, is your print Gargantua. Daumier clears his throat. Daumier: Ah yes, one of my finest caricatures. Presenter:

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