Pstnlavori

I...full functional with eCommerce capabilities where user can register , make payments and add balance into their accounts , add list of numbers and record their voice messages and run voice broadcasting campaigns, fax broadcasting campaigns and sms messaging campaigns, run surveys etc We need buisness partners that will be responsible for calls terminations through PSTN , sms termination through smpp with valid license to run the telemarketing services . We will be responsible for system functionality and our partner will be responsible for calls / sms termination and legalaties required to setup buisness , we like to work on 50% 50% profit / loss basis . We need professional proposal and will provide more de...

...format of the Japanese text, we do have special delivery requirements. This is because Japanese text does not wrap correctly on websites. The full details can be explained in detail later. You should have a background and/or experience in the telecommunications sector. PSTN and SIP. Our target market is business users, we do NOT provide consumer level services. INVITE Communications provides two (2) types of telecommunications services. The 'Multi-Carrier Aggregate Trunking' (MCAT) service combines PSTN/SIP from multiple carriers and delivers it as a single SIP connection. Mixing KDDI, NTT, NCOM, SoftBank, etc.. into a single line. The primary market for this product are Call Centers and businesses with a high call volume. The other product is...

Need to develop social app, which includes: live event management, IM chat and pictures exchange, calls/video between users, calls to PSTN numbers. Project will be developed in steps. More details will be given in private conversation.

I need a turnkey solution like Skype and must be able to make calls to GSM . DID, Calling card, PSTN terminations and mobile /PC dailer

1) Deployment of UCCE(Sprawler) 2)PSTN USER to Agent call Flow 3) Introduction to the monitoring tools 4) Trobleshooting on ingress gateway and the VXML gateway 5)CVP and ICM Logs

We need our GSM gateway that is connected to the Softswitch to transfer calls to the PSTN network. We have GSM gateway that receives calls from the Switchswitch from the Internet. We need an expert that will make the right configuration so that a call that is initiated through a softphone through a Softswitch will terminate a call through the GSM gateway.

Configure Grandstream HT503 as PSTN gateway to freepbx

I need a new website. I need you to design and build it. we are a group that holds voice only meetings everyday on skype which has a 25 person per call limit we would like to find a new platform possibly based on webrtc that has the pstn features skype provides us at a reasonable cost

- Set up a cluster with Elastix Mt (using kamailio, domains and 3 asterisks servers in load balanace) - The services must be configured in 4 VPS (1 proxy kamailio/ mysql; 3 asterisk) - The asterisks should automatically sync (config and all state) - The PSTN coming inbound calls will be sent to the proxy and should be sent to one of the asterisk with load balance. - Any asterisk should be able to transfer the call an extension that is connected to the cluster.

- Set up a cluster with Elastix Mt (using kamailio, domains and 3 asterisks servers in load balanace) - The services must be configured in 4 VPS (1 proxy kamailio/ mysql; 3 asterisk) - The asterisks should automatically sync (config and all state) - The PSTN coming inbound calls will be sent to the proxy and should be sent to one of the asterisk with load balance. - Any asterisk should be able to transfer the call an extension that is connected to the cluster.

The challenge is to have the Polycom 700 unit provide to the AMX 700 a calendar that shows the Skype meetings scheduled for the meeting room. From there, a user should be able to click on the meeting and launch a Skype meeting that will dial to a PSTN number associated with the Skype meeting, and allow the sharing of content from any of the computers connected.

VoIP features such as call waiting, caller ID, call forwarding and 911 services, match features available on traditional PSTN voice networks.

We want to integrate the possibilities of VoIP based calling to PSTN (landline) networks in our web app. Our need is a browser based softphone, like 'Zoiper' for desktop. Requirements: - input field for phone no. - call/hangup button - status massage box (calling, hungup etc.) - DTMF tone buttons (0-9 # *) I provide credentials for a sip server to handle the calls, including host, username and password. I have tried the one from , but it seems to be limited.

...(EXCLUYENTE) · Virtualización de servidores y desktops (VDI), configuración, puesta en marcha y administración de instancias virtuales sobre plataforma VMware (ESX 5.5) · Administración y mantenimiento de Storage EMC (EXCLUYENTE) · Administración de plataformas de Backup & Restore corporativas, Symantec Backup Exec y Beaam. · Administración y mantenimiento de redes (LAN & WAN), plataforma Cisco, MPLS, PSTN · Conocimientos de Firewalls Checkpoint · Conocimientos de PBX Avaya administración y configuración de teléfonos IP · Conocimientos de Telefonía IP – VoIP · Administración de herramientas de monitoreo. Competencias Humanas...

We have a 5 seater call center for which we need to setup the Epbx either Asterisk Based or any other Asterisk based software like Free PBX, Elastix etc. We have: 6 PSTN lines connected with FCTs having call Hunting facility. 8 Port FXO Gateway (Grandstream GXW4008) 24 Port FXS Gateway (Grandstream) A pentium Xeon Computer (Server). We have few requirements: 1) All Calls should be greeted with IVR (IVR Already Recorded) 2) If any agent is free, call should be land on his Computer/Telephone/Softphone 3) If all agents are busy, call should be placed in a queue, once any of agent get free, the call should land on their phone. 4) Call Logs and Recording Please contact, as its a urgent project and will be awarded soon. I am planning to close the project below 180...

We have a freepbx server that was just set up. I need our analog card set up so our incoming pstn lines work. I also need our enpoint manager set up so that our cisco 7960 phones connect and our digium d70 phones work. I have been working with another freelancer, but we are both lost and I need this done ASAP!

1. Setup the security and call quality settings correctly. 2. isymphony complete setup 3. incoming call diversion to operator logging, (may have 2 operators) 4. correctly setup/verify approx. 10 extensions and 2 pstn connection 5. call recording schedule into dropbox or Justhost folder 6. make sure pabx run smoothly without having any security issues 7. change ivr menu options and hold on music options 8. setup on demand recording individual files to as user option to email or downloaded 9. help to setup headset’s as optimum use conditions & rectify current issues with existing trunks 10. Given full tanning including; Free PABX commands essentials Setup and use of isymphony Setup new extentions call recording Managing backup’s

need to set up gxw4108 for termination from voip to pstn, i got 2 issues : 1- FAS while it is ringing it starts billing 2- want to hide the CLI which is the phone number

Hi I still need someone to develope my opensips server. I attached the project details What we have done: • Installed Opensips 2.1 with default configuration • Installed ASTPP • Installed RTPProxy • We are developing Android client We want you to do these jobs: • Solve NAT traversal problem • Integrate an installed ASTPP with Opensips • Pass SIP connectio...someone to develope my opensips server. I attached the project details What we have done: • Installed Opensips 2.1 with default configuration • Installed ASTPP • Installed RTPProxy • We are developing Android client We want you to do these jobs: • Solve NAT traversal problem • Integrate an installed ASTPP with Opensips • Pass SIP connectio...

we need a dailer app bid if you have already a chat app with video, audio, group chat, pstn calls, sms , g729 codec

See logs attachment first, pls When Asterisk IVR receives a transferred call from other Asterisk broadcasting dialer server ( Asterisk that makes calls to PSTN and when a callee answers dials "#1" ) , then IVR prompt audio is played but after 10 secs call cuts off. The objective is to fix cut off problem

Objective of project is to create a smooth callflow from AsteriskServer A to AsteriskServer B. I have two different Asterisk boxes: A) Broadcas...callflow from AsteriskServer A to AsteriskServer B. I have two different Asterisk boxes: A) Broadcast dialer B) IVR questionnaire I want calls dialed from A where callees dial " #1" to be transferred to B . Applications that performs dialing and IVR functions are fully developed, but is required to do some additional setups to ensure call flow goes from A to B. Currently when PSTN callee dials "#1" phone call cuts off before IVR (ServerB) is played. ------------ I will provide SHH access to both Servers and give you a quick overview about web apps that generate calls on Server A and for I...

I need a VOIP Server for maximum of 1500 simultaneous calls between Android clients and a PSTN network. I installed an Opensips 2.1 server on Ubuntu Server 14.04 but have these problems: 1. Opensips Clients can call each other but without audio transition. It seems that this is a NAT traversal problem. I installed RTPProxy but could not solve the problem. 2. How can I configure this server with PSTN network? 3. How can implement a prepaid billing for my Opensips using ASTPP? I installed ASTPP but could not connect it to Opensips. I want you to configure my server and learn me how can I do this myself.

Ich benötige einen 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 vorgegeben solange das Ziel erreicht wird. Die Aufgabe ist erfüllt wenn ein Nachweis erbracht ist, dass in 100 Bespielanrufen ( verteilt auf alle Mobilfunkanbieter O2, D1 und Vodafone ) das ziel im Durchschnitt erreicht wurde. Es handelt sich hier um eine hoch komplexe Aufgabenstellung, wenn Ihr nicht genau wisst was zu tun ist oder mir einfach eine SIPGATE Anbindung verkaufen w...

I need someone to help configure the Patton SmartNode 4112 (FXO Port) with Asterisk so we can use SmartNode 4112 as a VoIP gateway for PSTN lines.

... Configure AsteriskNOW with FreePBX to: Two (2) PSTN Lines (FXO) Two (2) Extensions (FSX) We would like that in case of Internet or Power fault, the PSTN lines route to FSX extensions Six (6) Extensions using Softphones on Windows and Android One (1) Extension using Ealink SIP-T21P Two (2) SIP Trunk services (i apreciate if you can recommend one) Configure outgoing routes to use the better phone costs on SIP Trunks. During the Work Hour, incoming phone calls on PSTN must play a Welcome Message and ring in some extensions at same time. In case of none of these extensions answer, ring another group of extensions. Out of Work Hour, incoming phone calls on PSTN must ring on another group o...

Analog USB Phone, which is a device that serves as an interface for connecting analog telephone line (PSTN / Landline) on the computer. Inbound or Outbound Call performed using a software phone (softphone). Users speak and hear the sound using the headset plugged in the computer. Expected Features : • Ring detector • DTMF Decoder/Encoder • Caller ID • Call Progress • Hook Detector • Flash • And other standard features telepony • Do not use additional power supply (enough from usb) • Telephone conversations recorded on the PC (Voice Record) We propose to use the SLIC AG2130 and Audio MCU Module HT82A834R as the main component. Product package consists of : 1. Schematic, PCB Layout & BOM 2. Source Code (Assemble...

I have a working Asterisk VOIP system ( Elastix). I have a Sangoma card with 4 PSTN lines installed. Caller ID from National Carrier is working on analogue phones. when connected to Sangoma Card it works 30% of the time, sometimes showing caller ID, sometimes not. it must be a setting in the conf files and card settings. I need someone who is familiary with those parameters to fix my issue.

I have a working Asterisk VOIP system ( Elastix). I have a Sangoma card with 4 PSTN lines installed. Caller ID from National Carrier is working on analogue phones. when connected to Sangoma Card it works 30% of the time, sometimes showing caller ID, sometimes not. it must be a setting in the conf files and card settings. I need someone who is familiary with those parameters to fix my issue.

... from-pstn default In Service 2 from-pstn default In Service 3 from-pstn default In Service 4 from-pstn default In Service 5 from-pstn default In Service 6 from-pstn default In Service ...

I have already a working Elastix box running my VOIP system and trunks. I have cisco SIP phones connected to the Elastix box and an FXO card for my PSTN lines. I use office 365 for my internal communication and emails. I need someone who is passionate about Elastix/Asterisk and knows how to integrate the different tools i mentioned above. so this project is about improving my current VOIP setup to allow the use of all the functionalities. my priority is to fix / implement: - caller ID not always showing on the trunk ( FXO card) - Fax integration with my email server - a shared address book ( LDAP?) that can be used by cisco phones or any soft SIP application connected to the box - integrate my office 365 lync ( now it is skype) with my elastix trunks - setup my VOIP sip prov...

Asterisk server configuration: We need someone to configure an asterisk server.. Here is the what we are looking for: 1. there will be 2-3 incoming lines (lateron we will move to PRI) 2. in asterisk system, one incoming number will be attached to 10 to 15 out going number. 3. incoming call will include a hash(#) and later digits as extension number 4. you need to make out going call to pstn/mobile according to extension number 5. Outgoing dial plan will have some logic based on time and other parameters Will provide detail requirement and scope of work after initial discussion

Need some assistance configuring 2 pstn lines from a fxo gateway to my asterisk

...Lync Server 2013 Front End (CORP-2012-LYNC1) 3. Lync Server 2013 Edge (CORP-2012-EDGE1) 4. Server 2012 ARR Reverse Proxy (CORP-2012-RP1) 5. Anynode Session Border Controller (CORP-2012-SBG1) The scope of this project is: 1. Configure Lync to utilize the Anynode SBC for PSTN connectivity. 2. Configure dial-in conferencing to utilize a DID delivered via SIP through the SBC. 3. Configure two users to make and receive calls via PSTN. 4. Place a test call from each user to the PSTN. 5. Receive a call from the PSTN to each user. Additional configuration in the Anynode SBC may be necessary. Support is available through the vendor. Connectivity will be available through VPN connectivity. The scope of this project must be completed in its en...

I did configured Elastix server to work with 4 PSTN lines (4 fxo card) and it was working fine ... However, I did add many extensions + IVR sounds ....etc. Then Elastix want answer my incoming. In addition, when I call it says all circuits are busy now. When I press 7777 I hear the main IVR but it won't answer any incoming calls! Please help

IVR - Install IVR system. Please provide your recommended software. I am...a number to speak to support You will be provided with detailed instructions once selected WebRTC - Install WebRTC software on our server - Create WebRTC system with the following use cases Use Case 1 1. User selects a contact from a list online and clicks audio call (simple html list) 2. System looks up the contact's phone number from the server 3. System calls the user by over a PSTN gateway. Must support calls to local and internation phone numbers 4. The caller and contact begin to talk 5. Call may be recorded Use Case 2 1. User selects a contact from a list online and clicks video call 2. Contact receives video call notification 3. User and contact begin video call 4. Call ...

We need advice from a person with experience in Kamailio deployment (including billing solutions / Failover) in order to set up our own SIP platform which will handle calls from our carrier grade Voip network to the PSTN via an SS7 gateway (already setup). Have a look at the diagram attached. Please only apply if you have experience with such deployments, thank you.

I have a SPA-3000 Linksys device which I want to use to: 1. Already have a voip account number 2. Need to route all incoming pstn calls to another number via VOIP

The task is to display a popup in a VOIP application when the call fails due to recipient not being connected to the server. The popup should give the user an option to call via the PSTN number instead, which is the username part of dialled SIP URI.

A VOIP application uses '408 Connection Refused' catch to end a call when the recipient is not connected to the SIP server. The task is to use this catch to display a popup before the call screen, giving the user an option to call via the PSTN number instead. The PSTN number is the username part of dialled SIP URI.

A VOIP application uses '408 Connection Refused' catch to end a call when the recipient is not connected to the SIP server. The task is to use this catch to display a popup before the call screen, giving the user an option to call via the PSTN number instead. The PSTN number is the username part of dialled SIP URI.

We are a bunch of developper who are working on a new website. So, we are able to do everything (front-end / design and all). We have to add webrtc capabilities to this website we are setting up. We want to have all the features that a digital PBX can give (web rtc call, video ...We are opened to Asterisk too. Here is what 1) api for webrtc extension/account management (crud) 2) api for list audio call recording 3) api for list envideo call recording 4) api for conference call 5) api for call history 6) api for voicemail box prompt management 7) api for working hour management 8) api for incomming call handling configuration 9) api for call forwarding to pstn or dont distrub feature 10) ivr for payment with credit card by phone. it will use http api supported payme...

I would like a System that has multiple operators accepting regular PSTN phone calls(Through any SIP provider of your choice) from customers who visit the website (Through click to call-Customer inputs their telephone number and Asterisk will place the outbound call; connecting them to the operator they selected.) Each operator will have a per minute rate(AsterCC or Star2Billing or another system of your liking) they can adjust themselves through their own Account section. The operators will have a dedicated webpage (Eg. ) while the customers will have an area to edit their account; but they will not have a dedicated webpage. The Asterisk server can be hosted by any provider you like, but the initial setup of these functions will be done by you. An extremely simple

we need a messenger app built for ios and android. This is a contest project . we need to have both done in 7 days sms signup chat group chat audio call video call pstn call only bid if you have a framework for server side and mobile side and all you do is change UI. WE NEED TO SEE A DEMO FROM YOU TO OFFER YOU THIS JOB AND YOUR DEVELOPRED APP MUST ALSO BE ON THE STORE AS PROOF.

We have a requirement to convert our existing SOAP API into a REST JSON based API. There are apparently software tools that can automate this in some cases. We use Java, Apache Tomcat, Hibernate, Spring, Microsoft SQL server on Microsoft server 2007, plus Linux as platform for all else. We provide global notifications via SMS, email, VoIP a, voice over PSTN (Asterisk based), fax and pager. All driven via web interface and email and test inbound initiations. Software stack diagram and more detailed brief available to best candidate(s).

We require Kamailio and Freeswitch Real Time integration. Implementation and Architecture should be well documented for us to manage the system. Architecture should include 1. N + 1 instance of Kamailio - Responsible for users registrations and user to user audio/video calls 2. N + 1 instance of Freeswitch server - responsible for voicemail, conferencing, pstn and media 3. Cgrates on Freeswitch for billing 4. Geo-Clustered MySQL DB for Kamailio/Freeswitch 5. DNS/Latency routing for Kamailio FE Please indicate reference jobs of this nature. Also skype to understand exact requirement

Need WebRTC Client and Server developer that will design and build Audio and Video communication from the Browser to Browser/Native Mobile Devices/Phones over different protocols including LTE, SIP, IMS and interface with PSTN. Client to Server must support Web Socket. I will look to you to provide recommendation on the server softphone but it must be OpenSource. Selected individual will also install WebRTC server software. Please note that this is not just providing JavaScript for WebRTC. This is client and server. The client will support the following: Works on Chrome, Firefox, IE, Safari, Opera, Bowser, Native IOS, Native Android Audio / Video call Screen/Desktop sharing from Browser to any SIP client Instant messaging Presence Call Hold / Resume Explicit Call transfer ...

Setting of small & medium sized networks with bus & star topologyrnCabling & crimping for systems, hubs, switches & routersrnOSI Layers and IEEE 802 standardsrnNetwork devices like Repeaters, Hubs, and Switches & Routers etc.rnTopologies, Media Access methods TCP/IP services, IP addressing & Sub nettingrnDHCP, DNS, Telnet etc. installation & configurationrnISDN, PSTN, security, Internet connection & sharingrnInstallation and Configuration of Windows NT, 2008 servers.rnConfiguring Access list, NAT, VLSMrnConfiguring RIP V1 & V2, EIGRP, IGRP, OSPFrnTroubleshooting Cisco 2800 series Routers. rn

Call termination PSTN setup using TDM packet injection expert required in VOIP in Pakistan pay will very exciting. contc. [The administrator removed this message for encouraging communication outside Freelancer.com, which breaches our Terms and Conditions - Section 13:Communication With Other Users.]

Setting of small & medium sized networks with bus & star topology Cabling & crimping for systems, hubs, switches & routers OSI Layers and IEEE 802 standards Network devices like Repeaters, Hubs, and Switches & Routers etc. Topologies, Media Access methods TCP/IP services, IP addressing & Sub netting DHCP, DNS, Telnet etc. installation & configuration ISDN, PSTN, security, Internet connection & sharing Installation and Configuration of Windows NT, 2008 servers. Configuring Access list, NAT, VLSM Configuring RIP V1 & V2, EIGRP, IGRP, OSPF Troubleshooting Cisco 2800 series Routers.