Trixbox h323 jobs

Full VoIP on flutter (Fonctionnalities) - Audio Call - Video Call - Instant Messenger - Interconnected With Magnus Billing user account - File transert - Video Recording -Audio Recording Codecs - G729 - G723 - G711 - GSM Protocol - SIP - H323 - TLS - ZRTP Platform - Android - IOS - Web

Need somebody who can configure a Cisco C60 Codec as Standalone device (No Video Communication server or similar) in order to join Webex meetings via SIP call. A free sip registrar like should be used, alternatives are appriciated. If possible there should be no reucuring cost in the solution (e.g. Hosted SIP registrar/H323 Gatekeeper like pexip) As deilvery there is either a complet configuration document expected or and lodable config file). A network configuration suggestion for the firewall should be proposed as well.

We need to build a WebRTC platform that can bridge H323/SIP videoconferencing systems. Must work either on premise or as SAAS and allow integration to other SW platfforms. Development in Angular is prefered.

I require a SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and freepbx. I need a SIP & Jitsi expert for: 1) create a sip/h323 connector for jitsi, 2) help me with getting an affordable india phone number for inbound connections – Twilio pricing is not affordable for Indian numbers 3) setting of sip trunk between jitsi and freepbx, so the participants from pstn can connect to the jitsi instance 4) Connect our jitsi instance to google transcription. I will provide the API key

The project is to create SIP/h323 connector for jitsi project and setting of sip trunk between jitsi and fusionpbx. Need a SIP & Jitsi expert to: 1) create a sip/h323 connector for jitsi, so, it can be given to hosts. 2) setting of sip trunk between jitsi and fusionpbx, so the participants from pstn can connect with jitsi instance deployed in our cloud.

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks

Have a Voip asterick software upgrade project Experienced voip PBX 3CX The Voip server is running trixbox software system that we require upgrading an install 3CX remotely 1. Backup the trixbox software system 2. Save all the current voip phone system configuration settings 3. Upgrade the voip system 4. Reinstall the voip phone configuration 5. Test the new upgraded phone system 6. Backup the new phone software system 7. Connect voip cordless phone 8. Connect mobiles phone to voip 9. Connect 3CX web chat to voip 10. Fix any minor bugs over next two months.

We are looking for a VoIP systems engineer to help complet...developed web based calling application linked to Asterisk/Free PBX. - Experience with Asterisk and/or FreePBX - Experience with Apache, CentOS, DNS, hosted services, MySQL - Network design – Working with Firewalls, DNS, Load Balancers - Experience in software as service architectures (SaaS) - General telephony understanding - Understanding of VOIP platforms like Trixbox, Elastix, Freeswitch or FusionPBX - Configuring various VOIP Phones and iOS/Android Smart Phones - Knowledgeable in IP Telephony, unified communications, data networking, telecommunications, video technologies and Call Center - Experience working with TCP/UDP, SIP, RTP/RTCP or other multimedia net...

A renowned organization is interested to work with freelance developers that have previous working experience with C / C++ and JAVA Web Development. We would be working together for a time period of six (6) months, during this time any required information (regarding the projects) shall be shared with the dev...developers that have previous working experience with C / C++ and JAVA Web Development. We would be working together for a time period of six (6) months, during this time any required information (regarding the projects) shall be shared with the developer. The project we will be working on is a Call Recording Software (Compatibility with Various PABX Systems, Cisco, Avaya, Panasonic etc.). Protocols: SIP H323 RTP TCPIP UDP Tool WinPcap Voice Decoder: GSM – G711 &ndash...

Designing and Implementing Cisco CUCM, CUC Clustering on UCS servers. H323, MGCP, CUBE implementation & Configuration.

im looking someone to install h.323 protocol on freepbx so i can use Avaya h323 phones with freepbx.

Hi metik, I noticed your profile and observed you answered on the past to a project when someone else wanted to interconnect one asterisk and one avaya system, using h323. I am on the same situation. Can you please give me some details about your experience here and how you can help me ?

We are looking for a technical person who has good telecom experience and can help us developing ringless voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : Http:// You can read about it and tell me if you can do it. We would ideally prefer some open source technologies like FreeSwitch trixbox etc.. Please bid only if you know about and able to help.

WE need to connect a cisco 7206VXR 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.

ccnp voice training and configuration for voice gateway mgcl sip and h323

I have setup a Trixbox CE server, now i want receive call that r forward thru ip to my server Just a small project, i have more work on the server later : *Custom scripts, for example, call from a list (excel or txt) automatic *Maintain server, Daily, weekly, add new trunks & routes *Security

Hi Olexander M., I hired you a while ago and need you to re do a job as we have had a new phone server installed. Here is the job you did before: I would like a simple wallboard program that will work with Trixbox. It would need to display three queues and three extensions. I want to display in real time how many calls have been answered in each queue and how many have been abandoned (a call going to voicemail is classed as abandoned)and also the hold time. Then I want to record the number of calls taken by each of the three extentions. There needs to be a way of resetting these totals to zero automatically at various time during the day and also a way of running a report over a longer period. I have mocked up a design of what I'm looking to end up with.

I need to get done small job to configure Cisco 1861 router for fxo and fxs to work with elastic or trixbox open source. Network Designed as GPON 100 and vdsl which is working in redundant fail over in Mikrotik and port forwarding coz dynamic ip. GPON + VDSL MIKROTIK SWITCH 3750G AND two more switches 2960 and cisco phone 7960 its a quick job only configration need to done all network is already and runing engineer need to do 1- Elastix configure 2- cisco 1861 router for FXO AND FXS ports 3- Cisco 7960 Phone to configure to work with elastix Only experience guy who can done it right away Br Fiaz

I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that provides ringless VM : Http:// You can read about it and tell me if you can do it. We would ideally prefer some open source technologies to be used with it like FreeSwitch trixbox etc.. Please bid only if you know about and able to help. Thanks

...Server Server Y = Window based Asterisk Client server Main goal minimize Bandwidth in client side with quality voice . Required bandwidth compression (upto 60-80% in reality then theory) from Server X to Server Y. A usual SIP in G729 call takes 27-32kbps per port Explanation of scenario: 1. Server X (asterisk server, with static IP) receiving VoIP calls from different Carrier/Originator, with h323/sip protocol, using G711,G729 and/or G723r6.3 codec and sending calls to Server Y. 2. Server Y {Windows based Asterisk server with PRIVATE NETWORK IP, receiving calls from server X and sending to gateways (quintum, goip etc brand) or E1 cards. (Please take a look at this page for better clarity ) 3. Number of Server Y can be unlimited

Install and configure freepbx with inbound and outbound dialpaths setup including an IVR for receptionist. Any IP-PBX with ease of use and interoperability with sip and h323 devices will be good. Thanks

Mobile App for All Main Devices and Operative Systems Apple IOs Android Blackberry Windows Phone Customer Experience* Ability to Make and Receive Calls Balanc...Success Ratio) CDR (Call Detail Record) Country/City Reports Customer Reports Real-Time Monitoring Tool Security Logs Unsafe Account Monitoring Call Management DID Forwarding DID2IP Direct DID DTMF Support via RFC 2833 Hunt-Stop & Route Suspension IP Video Support Least Cost Routing (LCR) Multi-Dial Plan ystem Management Advanced RTP Management Auto-Provisioning Class 5 Server H323 Support Redundancy SIP Card Spanish Platform Support Ticketing System Billing Auto Recharge/Recurring Billing Disconnect Fee Invoicing Minute Shrinking Package Billing Step Billing Uni...

Need to make changes to extensions and multi-ring settings on various extensions for very old system with polycomm phones

...features normally found in a mobile dialer (example: viber,vonage,vippie etc). c. Following features must be available in the dialer: Making & Receiving VoIP Calls Over 3G,4G, WiFi, GPRS. -Video calling -Voice calling -Messaging -Filesharing dialer should also provide Vpn/tunneling support for working in SIP/Voip blocked countries. Dialer should support SIP/H323 for signaling and should work with any standard SIP server. Dialer supports G729, G723.1 and codec for sending audio stream. Dialer can run behind NAT or on private IP with VPN Tunneling embedded. with phonebook contacts , Hang up, Redial,Call Hold,Call Mute,Call Forwarding,Call Waiting,Speaker Phone,Keypad Display,Call History: All, Recent, Dialed

...features normally found in a mobile dialer (example: viber,vonage,vippie etc). c. Following features must be available in the dialer: Making & Receiving VoIP Calls Over 3G,4G, WiFi, GPRS. -Video calling -Voice calling -Messaging -Filesharing dialer should also provide Vpn/tunneling support for working in SIP/Voip blocked countries. Dialer should support SIP/H323 for signaling and should work with any standard SIP server. Dialer supports G729, G723.1 and codec for sending audio stream. Dialer can run behind NAT or on private IP with VPN Tunneling embedded. with phonebook contacts , Hang up, Redial,Call Hold,Call Mute,Call Forwarding,Call Waiting,Speaker Phone,Keypad Display,Call History: All, Recent, Dialed

The task is to develop very simple VoIP application using C++ and Opal Library which must do following: 1. Must accept calls H323 and SIP 2. Possibility to set PDD/SCD time 3. Possiblity to set time and continuously play WAV file. Please apply only if you have experience developing VoIP application!

Hello! I need DID telephone number of Macedonia (code +389) for a short period of time (about 10 days). Incoming calls are planned only. No originating. Suitable prorocols: IAX2, SIP, H323. At my side - Asterisk PBX v.13 Thanks.

I need Avaya 1608 H323 EXTENSION TO WORK WITH ASTERISK Free PBX

Configure asterisk to take calls h323 and send it as SIP.

Convert Trixbox 2.6 to latest FreePBX, convert SIP-Trunk, 10 extensions and Routes

...tunneling (Relay Based) We are looking for a mobile dialer(Full source code) SIP or H323 (SIP is preffered) which can run on 1) User friendly interface 2) Mobile Dialer should support SIP/H323 for signalling. 3) Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded (Relay Based) 5) This software will run on the phone having iPhone, Android Symbian or Windows Mobile 5 and 6 having internet via GPRS/EDGE/Wi-Fi/3G and all other available OS and Phone models. 6) Mobile Dialer should be fully customization. 7) Able to use Mobile Dialer with any kind of Softswitch,Voip Billing Platform which support SIP/H323. (VOS, VoIP Switch) The dialer must be runing from ...

My trixbox died, I must rebuild it. I need someone familiar with setting up a T1 card and possible some help with getting it functioning on a vlan. My main network is on vlan1, my phones were on vlan2. This was all setup before my time. I have trixbox up, but I do not know how to setup the t1 card. Please let me know if you can help.

I am seeking c++ developer to get phone number from H323 package. I am getting the H323 package with Winpcap. The small tools will be run windows OS. If you can do it , please bid. Best Regards wei.

Looking to have someone create a simple script or website where a user can click a button from a device to test video. Something similar to speedtest, but will need to have an output of different file sizes that also include the quality of network data, such as jitter , 720HD vs 420HD , etc. Report should be detailed from a network admin perspective. It should also have a config of some sorts so you can change the testing server etc.

Telecom • Cellular Networks, GSM, GPRS, 3G, 4G. Network • LAN, WLAN, WAN. • OLSR, RIPv1/v2, UDP, TCP/IP, DHCP, DNS, DynDNS, DNS Forwarder, AD, GPO, VoIP, SIP, H323, WDS, Routage, NAT, Proxy, FreeRadius, IDS/IPS, Firewall, HA (High Availability), Load Balancing, Captive portal, DMZ, VPN, VLAN, Ipsec, OpenVPN, troubleshooting. • Cisco (Routeurs, Switchs), CCNA. Systems • Windows (2003, 2008) Server, XP, Vista, W7, W8, GNU/Linux (Gentoo, Debian, RedHat, Ubuntu, BackTrack, Kali), FreeBSD (pfSense, OpenSense) Tools • Security: PKI, Cryptographie, Metasploit, Nmap, Nikto, IPTables, Kismet, Snort, SSL, WEP, WPA, WPA 2, Honeyd, Honeypot, Nessus, Antivirus, Volatility. Burpsuite, OllyDbg, IDA. • Networks: Packet Tracer, NS2, GNS3, Squid, Squidgu...

Hello, Currently Use TrixBox & Google Voice, Asterisk, for Our VoIP. We cannot get the IVR working, I am sure this is a simple line fix but cannot figure it out Thanks

I need help from someone to setup a predictive dialer. Will start immediately. Goautodial, trixbox or Vicidial.

1) What is dot1q protocol ? 2) what is HSRP , VRRP, GLBP? 3) what is vtp and various modes of VTP 4) voice h323 something like this protocol? 5) RFC for private and public addressing space ? 6) class of service and QoS ? 7) states of BGP 8) how to see port status in catalyst ? 9) how ospf priority determined ?

We have a trixbox and we had our fax to email working but suddenly stopped working. I need help getting his back up and running.

I need a plataform based on Opensource implementation (like Asterisk, Yate, etc.) that implement a simple H323 to SIp gateway with video support. The project consists of: 1. Elaborate the solution 2. Install remotly the solution on my server 3. Document the solution, providing details what's components were used to make it, and what's implemented.

H323 TO SIP VIDEO CALL ON ASTERISK

I am looking for Script/Configuration of our Trixbox system to make changes to an out-pulsed caller ID number (adding a suffix) Case: A caller calls a queue and their number for example is 655-555-1212 The PBX will ring agents at their extensions or forwarded numbers. The agent's caller id display will display the caller's number and adding a suffix to it. To identify that the call is from the Trixbox and not a telemarketing call. Example of number to show: 655-555-1212-999 so we want to add a 3 digit number at the end of the out-pulsed caller ID. This should be a simple thing to do for a developer that have experience with VOIP PBX systems.

1) backup existing dedicated server installation 2) setup of a new VPN 3) adding currently installed vos3000 softswitch to VPN 4) adding voip-gateway (linux) to VPN to receive voip-calls via VPN (h323) 5) voip-gateway has single network interface, preconfigured for VPN use in previous project; needs to be configured to new VPN 6) configuration vos3000 for signalling and RTP/media-routing via VPN 7) backup again once all successfully tested required skills: vos3000, linux, vpn, voip-routing, network configurations

I need the following steps: 1) Setup new dedicated server and install virtualbox 2) backup existing installation of VOS3000 and restore to virtual box server incl configuration to new IP 3) setting up virtual private network VPN server to another virtualbox server 4) configuring voip/RTP routing from VOS3000 through VPN, via Asterisk installation in virtualbox server 5) configuring VPN...dedicated server and install virtualbox 2) backup existing installation of VOS3000 and restore to virtual box server incl configuration to new IP 3) setting up virtual private network VPN server to another virtualbox server 4) configuring voip/RTP routing from VOS3000 through VPN, via Asterisk installation in virtualbox server 5) configuring VPN and RTP routing (ports) in receiving voip server (...

We have 3 offices with a trixbox in each. One died, trying to rebuild it. I need someone familiar with setting up a T1 card, site to site trunks. I can handle the extensions and phones.

...Efforts by our existing engineer to date but have not resulted in a solution: ------------------------------ - Previous efforts on 8/21 (reset phone and rebuild tftp boot file) recorded in 234543 - PBX Status console shows x1168 registered and online at , which matches the configured device for TFTP boot file, no status differences from other extensions and devices - Verify Asterisk/Trixbox configuration for Extension, Inbound Route,  TFTP boot settings specific to phone/extension, no issues noted, effectively identical to numerous other working extensions/phones except for extension number and display names - Examine all custom extension, follow-me, ZAP channel DID, Day/Night control, IVR, Queue, Ring Group, Time Condition/Group, Conference and general settings, no referenc...

...Please visit or we are looking similar product. OS: Symbian, Windows Mobile 1) User friendly interface 2) Mobile Dialer should support SIP/H323 for signalling. 3) Mobile Dialer supports G729, G723.1 and G.711 codec for sending audio stream 4) Mobile Dialer can run behind NAT or on private IP with VPN Tunneling embedded 5) This software will run on the phone having Symbian or Windows Mobile 5 and 6 having internet via GPRS/EDGE/Wi-Fi/ 3G ... 6) Mobile Dialer should be fully customizable. 7) Able to use Mobile Dialer with any kind of Softswitch,Voip Billing Platform which support SIP/H323. 8) Able to voice / chat with Gtalk, Yahoo Messenger, Skype, MSN, and popular social networking. Please Do not ask for any prepayment Until Project is been tested and