Inform opensips pstn gateway jobs

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    987 inform opensips pstn gateway Jobs gefunden, Preisgestaltung in EUR

    ...Spezialisten im Bereich Telefonie im PSTN Netz. Die Aufgabenstellung lautet, eine Verbindung zu einem Gesprächspartner in 500 ms - 800 ms herzustellen ( es zählt die Zeit bis die Leitung besetzt ist ). Die Gegenstelle ist in 90 % der Fälle ein GSM Teilnehmer. Wie die Verbindung Aufgebaut wird ( z.B. Asterisk mit SIP Gateway oder GSM Dongles ) wird nicht

    €6787 (Avg Bid)
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    2 Angebote

    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.

    €68 (Avg Bid)
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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

    €24 / hr (Avg Bid)
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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

    €144 (Avg Bid)
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    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

    €67 (Avg Bid)
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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

    €635 (Avg Bid)
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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

    €345 (Avg Bid)
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    ...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....

    €218 (Avg Bid)
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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......)

    €21 (Avg Bid)
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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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    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

    €1113 (Avg Bid)
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    Hi. I want to setup kazoo from source such as contact center and config to perform outbound, inbound call with PSTN network and softphone. And have some other requirements. Can you do it?

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    ...State.. 4. The System will handle Prepaid Calling Cards allowing them to top op their accounts for DID usage and termination services. 5. Use Sip Router (Kamailio, Openser, Opensips) Load Balancing, Registrar, Routing outgoing calls by LCR for termination 6. Integrate a billing system (A2Billing or suggest other). If A2 Billing, code for email dunning

    €1274 (Avg Bid)
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    ...forwarded to a cellphone. I use a Grandstream Gateway (GXW4104) for 3 PSTN lines. The Gateway is accessible but for some reason it stopped working for 1 week now. I tried different things but it's not dialing out. I added temporarly a SIP trunk so the systems continues to work. I need to get this Gateway working again. I can provide a Team Viewer access

    €80 (Avg Bid)
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    se trata de dos centrales de voip las mismas una esta en amazon como un contenedo...resolver la comunicacion de punta a punta entre ambas centrales, los flujos buscados son: PSTN > FXO > Grandstream > Raspberry > Asterisk en la Nube > Raspberry > FXS > Telefono Telefono > FXS > Grandsteam > Raspberry> Asterisk en la nube > Raspberry > FXO >

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    Simple SIP redirect Routing system ,listen on port 5060 .It authenticate the incoming SIP invite source IP address , retrieve data by connecting to Mysql or Sql server, get values that should be appended to sip contact header edit/append sip contact : content ,then send it back to source of invite ,wait for ACK and send invite again if no ACK within 3 seconds (max retry 5 ). 2-Installat...

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    Asteriks Transfer Dialplan Beendet left
    VERIFIZIERT

    I have a running pure asterisk sytsem. I need somebody to write an asterisk transfer dialplan. I need to transfer the PSTN call landed in one zoiper extension to another PSTN Number.

    €33 (Avg Bid)
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    Hi I want to modify Linphone into Tablet 7" Andriod 5.1 use for Free SIP to PSTN Kiosk [Zur Anzeige der URL Anmelden] - Auto open app in fullscreen mode (portrait) without close button or switch to other app after device power on - Full screen vdo/jpeg/gif file from URL List - after any touch from above will show

    €485 (Avg Bid)
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    Free PABX error fix and PSTN Gateway setup ones it fixed transfer 2 x Free PABX from digital ocean droplet to Linode Setup maximum security or firewall call recording setup to cloud drive

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    Kazoo konami transfer via api for mobile extension. _incoming call comes from PSTN and is connected to mobile phone via PSTN -we want to find the channel from kazoo using konami and transfer the call, by sending command via mobile app. WOuld like to have help with konami, in order to transfer the call.

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    I am using HT503 to automate calls to PSTN. I am not sure what settings to use for this to work in India. I am able to make calls from one sip to another, so that's not an issue. I have also successfully connected FXO of ht503 to asterisk(Registered). This is the output I see everytime I try to make a call: > Event: Hangup > Privilege: call,all

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    I need a script in [Zur Anzeige der URL Anmelden], [Zur Anzeige der URL Anmelden] and Laravel 5.6 which informs users that the message is being typed. I am not using Pusher and redis.

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    OpenSIPS/Sippy Carrier Routing - Stage 1 either an OpenSIPS or Sippy box will act as an outbound traffic router for multiple carriers. Calls will be sent from already running Asterisk boxes to the OpenSIPS/Sippy/SippyGO box (we will prepare base OS, OpenSIPs/SIPPY core installs) Media needs to be proxied with RTPProxy/etc through the OpenSIPS/Sippy/SippyGO

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    ...notification in opensips. We used new module which provide push notification, but I think there should be other way to wake up the linphone app, LOGIC: some way linphone app will send push device token to opensips... opensips will fetch device token and store it on database for future use, when app will try to call to linphone app opensips will check that

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    [Zur Anzeige der URL Anmelden] I prefer new freelancer.

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    Need someone with openSIPS/freeswitch experience the initial task would be to help us setup and configure a high availability deploy of OpenSIPS ->[ Freeswitch1, Freeswitch2, Freeswitch3, … ] the other aspect is I want to setup OpenSIPS to handle … - [Zur Anzeige der URL Anmelden] - [Zur Anzeige der URL Anmelden] Where each subdomain has different...

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    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

    €172 (Avg Bid)
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    ...Pointing number to PSTN --------------- - CREATE TRUNK ([Zur Anzeige der URL Anmelden]), as in documentation provided 'Sample 2', create PSTN forwarding to your defined number. - UPDATE DID ([Zur Anzeige der URL Anmelden]), referring to 'Sample 2' you should assign one step back created PSTN trunk to your

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    ...Pointing number to PSTN --------------- - CREATE TRUNK ([Zur Anzeige der URL Anmelden]), as in documentation provided 'Sample 2', create PSTN forwarding to your defined number. - UPDATE DID ([Zur Anzeige der URL Anmelden]), referring to 'Sample 2' you should assign one step back created PSTN trunk to your

    €42 (Avg Bid)
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    Hello, We are looking for someone for long term projects with expertise in freePBX. We have freePBX account with plugins. We wan...Setup 3 DID providers Setup freepbx to be able to email PIN to users free PBX to identify caller based on CLI freepbx to connect participant by dialing out ( charging user pstn rate fax-to-email email-to-fax

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    ...an old PABX telephone system that is playing up. I've bought a second hand system that I would like installed in its place. The business its getting installed in has 3 or 4 pstn lines coming in that are used for calls with line hunt, fax, internet and chubb security. The lines are currently playing up so it might be the pabx or some of the joins

    €22 (Avg Bid)
    Lokal
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    1 Angebote

    ...an old PABX telephone system that is playing up. I've bought a second hand system that I would like installed in its place. The business its getting installed in has 3 or 4 pstn lines coming in that are used for calls with line hunt, fax, internet and chubb security. The lines are currently playing up so it might be the pabx or some of the joins

    €108 (Avg Bid)
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    6 Angebote

    ...am looking for a voip softswitch with LCR and carrier failover. and billing of some sort. Possibly I can just use CDRs to calculate the billing. My plan was to use either opensips or kamailio I need two instances. one is around 3000 sessions and 100cps the other one is 7000 sessions and 250 cps but It would be nice to allow it to go to 12000 sessions

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    Hello, I need opensips server installed in front of our a2billing server so that we can have a load balancer to get more channels out of a2billing. All our servers are hosted virtual machines

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    Needing a VOIP softswitch for inbound and outbound calls. Need LCR. Was thinking opensips, CDR stats. I can build my own dashboard to query CDRs for billing.

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    Hi there. We are a new tourism company in Chile, we do tours to all around Chile, our target is the European market. This project consists of creating all our content to get started such as blog articles and sales pages, very important, they need to optimize for SEO. The articles need to inspire the reader, need to make them feel like to come to an adventure in Chile and, at the same time, it ne...

    €17 / hr (Avg Bid)
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    ASTPP Fix errors Beendet left

    ...Database connection fail...!!! 2018-01-13 13:00:25.568487 [DEBUG] [Zur Anzeige der URL Anmelden] [ASTPP] [LOAD_CONF] Query :SELECT name,value FROM system WHERE group_title IN ('global','opensips','callingcard') 2018-01-13 13:00:25.568487 [ERR] [Zur Anzeige der URL Anmelden] DBH NOT Connected. 2018-01-13 13:00:25.568487 [ERR] [Zur Anzeige der URL Anmelden]...

    €121 (Avg Bid)
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    Need an expert with hands on experience in deploying freepbx and opensips on cloud for architecture discussion and documentation. Deployment itself will be another project.

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    Chat App with PSTN/ GSM calling capabilities To be able to - Call PSTN lines - Call in-app (peer 2 peer) - Voice Notes - Instant Chats Media SDK IOS/ Android/Web Media SDK, integrated via griddle to android app and via Cocoa Pods to iOS app to enable communication between mobile applications or web for desktop to PSTN. Integration of

    €602 (Avg Bid)
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    ...termination services. Use Sip Router (Kamailio, Openser, Opensips) Load Balancing, Registrar, Routing outgoing calls by LCR for termination Integrate a billing system (A2Billing or suggest other) Allow scalability Integrate with CRM To be installed Sip Routers (Kamailio, Openser or OpenSips ) Split for incoming and outgoing calls Asterisk Realtime

    €2580 - €4300
    Featured Versiegelt
    €2580 - €4300
    24 Angebote

    Need opensips installation and configuration ip hiding, rtpproxy, pike modul conf., cgrates, multitenant users and groups, load balance with media(fs), class 4 features. Urgent delivery will paid extra tip.

    €172 (Avg Bid)
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    Have to read files in inform 7.

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    ...SDK IOS/ Android/Web media SDK, integrated via gradle to android app and via Cocoa Pods to iOS app to enable communication between mobile applications or web for desktop to PSTN. Integration of the media SDK will be done by the customer or contracted as professional services. media platform (hosted) media platform that enables users to communicate

    €255 (Avg Bid)
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    ...call only PSTN Number : again here you will need to send to our server - the number user want to call - login/password account of the user (this login/password is already stored in the app because user need to set it up to use the app) - finally a gateway ID (user will choose to initiate the call by choosing a given gateway id. Ids of gateway will be

    €577 (Avg Bid)
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    Need to troubleshoot some issues with opensips.

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    I am looking for someone that connects my Twilio numbers with my Xlite or Bria softphone. As far as I understand, a SIP registrar is being needed and OpenSIPs seems to be an appropriate solution. I am open for alternatives. A server is here.

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    3 Angebote

    Hi , I am looking for VOIP switch developers for calls routing on apps and PSTN

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    Deskside Services Beendet left

    ...with Apple Mac and IOS devices - Support Multifunctional Devices (MFD) for issues like Scan to Email, Scan to Fax, Email to Fax etc. - Support In-country PBX networks and PSTN interfaces - Support Hardware/Software selection and Procurement effort - Support Hardware Refresh, Redeployment and Disposal activities - Troubleshooting and resolving software

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