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    1,014 inform opensips pstn gateway Gefundene Jobs, Preise 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

    €6935 (Avg Bid)
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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

    €139 (Avg Bid)
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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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    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 [Zur Anzeige der URL Anmelden]

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    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 [Zur Anzeige der URL Anmelden] 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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    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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    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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    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 Beendet 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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    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

    €183 (Avg Bid)
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    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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    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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    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" Beendet 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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    Featured Garantiert
    €80
    251 Einträge

    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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    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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    create voip application working on PSTN and voip service, in which we generate virtual number, for example - [Zur Anzeige der URL Anmelden] - [Zur Anzeige der URL Anmelden] - [Zur Anzeige der URL Anmelden]

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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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    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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    ...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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    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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    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

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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

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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.

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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

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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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