Inform opensips pstn gatewayработы

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    1,023 inform opensips pstn gateway работ(-а,-ы) найдено, цены указаны в USD
    opensips + rtpproxy + asterisk Завершено left

    opensips (или что то другое) смотрит в мир исходя из домена на какой идет подключение(смотрит в базе mysql) направляет на нужный asterisk в локальной сети. нужно настроить opensips + rtpproxy

    $200 (Avg Bid)
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    1 ставки
    ASTPP OpenSIPS integration Завершено left

    Hello, We have ASTPP 3.6, having wanted integrated OpenSIPS. When we activate the latter, we can neither add or list the OpenSIPS devices. Only dispatcher is working in add or list. Thank you,

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    I need to make home page in magento to inform customers to install our app to use our services and inform cutomers website is not supporting orderings if he need to to buy somthing he need to download our app

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    Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls

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    Im setting up a network for VoIP. Am looking for an engineer that can assist in setting up the network with Opensip as the SBC and loadbalancing to 2or more Freeswitch servers. Security and firewall is a must in the network. Must have the ability to design the network. All equipment will be in the datacentre.

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    Setup a reset NEC SV9100 PBX with two phones and access to a single PSTN Australian Telstra Line System has NEC licences already applied. PBX Contains Blades: GCD-CP10 GCD-4COTC GCD-8DLCS GCD-4LCF PHONE are: DT400 and DT800 Project requires a tech with intimate knowledge of NEC SV9100 and DT400/800 devices.

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    Configure OPENSIPS Завершено left

    1: Functional characteristics - Individual RTP media streams shall use an associated RTCP channel to report transmission performance. - The media transported in 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....

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    I have 1 OpenSIPS Server, 1 A2billing Server and 2 Asterisk server setup in a working environment, I want to add another Asterisk server to the setup, Asterisk 13 is installed on Ubuntu 18 Server and ready to configured, I am looking for someone to add this server to the setup! If awarded this person should start right away.

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    I have a HID based USB caller id device which i want to use to get caller id on my pstn and inject the callers number and date and time in mysql database . If you can do that please bid this will be only on centos7 linux environment and it needs to run 24/7 on the server if it fails anytime it needs to try and reconnect or send failur email to admin/root

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    we are planning to introduce voice based platform in Google home and Amazon echo gadgets for customers investing in stock market.

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    WE need to connect a cisco 72...with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network.

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    ...It works fine. However its flakey I have a Cisco 3945 with Voice/UCS license that I'd like to run a SIP server on IOS to replace this. I only use this for LAN, there is no PSTN calling. I need my LAN devices to register to the SIP server and also my 4G Mobile device (using Zoipher client) when it's not at home (it uses a Split DNS entry to register

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    Program Webhooks in LaML Завершено left

    I need a programmer that is familiar with webooks and the LaML language. I need DIDs programmed to ring a SIP URI and or forward to a PSTN line and SMS/Text redirected to another number. All programming needs to be done on [войдите, чтобы посмотреть URL]

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    install and configure freepbx Завершено left

    need freepbx install on an inhouse server. have sip trunk and pstn lines. need configuration for IVR. 4 auto-attendants. simple call flow. need call monitoing, call recording and SMS

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    Configurar um server opensips/kamailio para controle de ddrs,chamadas para gateways externos , chamadas entre ramais , cdr com interface grafica.

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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

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    in Vicidial, I have two goip boxes, the first ...and [войдите, чтобы посмотреть URL] that I use for predictive dialing with normal agents. right now I transfer the affirmative calls from the survey 100.99 to the fiscal agents in 100.100 via PSTn so I want to pass them directly from one server to the next one I have already make the trunk for each one see each other

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    1- We want my VOIP app to 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play. 3- Give me source code after finish the job.

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    ...Account management Gateway management Routing gateway Mapping Gateway Settlement account user management Dial Plan CDR #You have to create everyday a new CDR Table i need a CDR page where i can check all the calls report like cdr by account name/gateway name or ip/client name or ip/ make sure i can use a SIP account as a routing gateway I want to make

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    UC540 PSTN troubleshooting Завершено left

    I have a uc540 with a spectrum pstn connected. Calls cannot be made coming in to that number. I called spectrum and I was told they are showing its busy.

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    1- We want my VOIP app to 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play.

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    6 ставки

    1- We want to use 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play.

    $26 (Avg Bid)
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    2 ставки

    1- We want to use 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase. 2- Upload to google play.

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    PSTN design Завершено left

    We have a PSTN design that is failing one test for China NAL testing. It passes in all other countries. The only failure is Return Loss at 2km, and it is failing by 0.5 dB We use the Silicon labs Si3050 and Si3019 chips and their standard reference design for their DAA We are looking for someone who direct experience using these ICs and the SiLabs reference

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    PSTN design - 09/09/2018 10:29 EDT Завершено left

    We have a PSTN design that is failing one test for China NAL testing. It passes in all other countries. The only failure is Return Loss at 2km, and it is failing by 0.5 dB We use the Silicon labs Si3050 and Si3019 chips and their standard reference design for their DAA We are looking for someone who direct experience using these ICs and the SiLabs reference

    $208 (Avg Bid)
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    1 ставки

    We want to use 1- Keep Awake and 2- Push Notification always on even app is closed in opensips for android first. Must be familiar with linphone, opensips and firebase.

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    2 ставки

    We want to use push notification in opensips for android first. Must be familiar with linphone, opensips and firebase.

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    We want to use push notification in opensips for both ios and android. Must be familiar with linphone and opensips.

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    We want to use push notification in opensips for both ios and android.

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    I'd like to write a message composed of 3-4 sentences to inform my management that I passed a Professional Engineering Exam. The message should be professional and powerful. Look at the attached file that will show you the expected format and content of the message. thanks.

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    ...I have CME 11.5 installed on 4331 router, I have configured the auto attendant and it's working perfectly when testing from inside the network, but when I call from outside (PSTN) I hear the welcome message but I can't enter any extension, like when it says press 1 for sales or 0 for help, it doesn't matter what I choose, I keep pressing 1 and 0 but

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    Project for Ruben R. Завершено left

    Hi planetelecom, I noticed your profile and would like to offer you my project. We are wanting to build a PBX for Hilton using opensips.

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    Trophy icon Design a Logo for "Virtuo" Завершено left

    ...manufacturer and vendor of the well-known carrier-grade Mediatrix gateways, is a global trustworthy partner supplying VoIP gateways for the Telecom industry focusing on SIP Trunking, PSTN/TDM replacements, Unified Communications, and Hosted Services. Its portfolio of VoIP gateways allows businesses to implement and manage reliable, robust, secure, cost-effective

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    Избранный Гарантированный

    I would like to ask you install some really good plugin that inform users that a site uses cookies and to comply with the EU cookie law GDPR regulations. The site for american users, but there are european people too. You need explain your choosing and install it to Wordpress Version 4.9.7.

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    build me a website and app that look like tinder, i will inform you the detail once you got the offer

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    build me a website and app that look like tinder, i will inform you the detail once you got the offer

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    Help with OpenSIPS Getting Started Завершено left

    I need help in setting up an OpenSIPS server and creating a SIP Proxy that alters some headers. I have knowledge of VoIP and SIP, but no experience in OpenSIPS. I'm a professional developer and do systems administration so i should be able to learn quickly. I would like someone who can teach me the basics, and prove me with answers to questions i

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    Support VOIP and PSTN Calls Завершено left

    create voip application working on PSTN and voip service, in which we generate virtual number, for example - [войдите, чтобы посмотреть URL] - [войдите, чтобы посмотреть URL] - [войдите, чтобы посмотреть URL]

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    I ...A200 card and a single PSTN connection on Port 1. What I need to do is the following: 1. For all inbound calls to the PSTN line, I’d like them routed from BDQ-PBX to PHX-PBX to the Main IVR; and 2. For all outbound calls from PHX-PBX that start with 01191, I’d like for those calls to be routed as outbound dials to the PSTN line on BDQ-PBX.

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    ...connect a Client to a Carrier that is scalable. However, we do not want to disclose any of the Carrier IPs to the Client and Vice Versa. I've done some research and using OpenSIPS and RTPproxy can help with this but I am having trouble setting this up on AWS. For the proof of concept, we can use 2 PABXs one to act as a Carrier, another to Act as a Client

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    ...someone to complete the initial configuration of an audiocodes mediant VE box ASAP. Initial Configuration is for 2 x IP-IP connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration

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    install FreePBX v.13.0.121+; Завершено left

    1. Install FreePBX v.13.0.121+ in to Linode server 2. configure SIP trunk engin (Australia) , (can be copy from existing PABX) 3. Connect PSTN to sip using ATA adaptor 4 Connect with Britex 24

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    ...am not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers

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    Can you make Linphone wake up with flexisip or opensips or freeswitch push notification and receive incoming calls or sms ? Freeswitch and linphone is already up and running my budget is $500

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    freeswitch project Завершено left

    ...message. All calls must be recorded. I need to be able to query CDR to check all communication between parties + open recorded communications. Incoming calls will come from PSTN, SIP trunk, and all outgoing call through sip trunk. Script may be based on freeswitch as it’s an xml based solution, faster for administration....

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    i am working on an app that can provide unlimited calling from Canada to India( same ...on an app that can provide unlimited calling from Canada to India( same functionality as Rebtel). I looking for someone who can provide pstn gateway solution in Canada with unlimited incoming. Any type of pstn gateway solution can be considered( ex. pri, sip......)

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    We are building a freepbx server in Canada that can receive inward and outward calls and need some advising setting up a pstn gateway. For example pri line would be better or a sip trunk or if there is any other option ? Looking for a knowledgeable person who can get this project started. p.s. we will be posting some more jobs for the same so will

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    usb caller id for android device Завершено left

    use usb modem or an artech AD102 to gain caller id on android device - the modem will be connected via USB I just need that done You have to do your own research ...have to do your own research and develop I have access to Artech AD102 and a usb voice fax modem to test with which will connect to my android device This will be used in UK pstn line

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    Hi I would like to configure Kamailio or OpenSips for load balancing of some freeswitch servers and I would like to use ASTPP as billing for that system. I would like to have about 1500cc Thank you

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