Download startrinity softswitch

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Download StarTrinity Softswitch latest version for Windows free. StarTrinity Softswitch latest update: Septem Download StarTrinity Softswitch latest version for Windows free. StarTrinity Softswitch latest update: Septem

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Reads some specific list of numbers, check the number is dialable or not. Save report to file. If call is answered, drop the call immediately. Also drop the call when got "ringing" signal. OTT bypass, other international bypass fraud detection Looped traffic generation ---> Sample CallXML Scripts for StarTrinity Softswitch Simple IVR-based filter Play IVR files 0.wav, 2.wav, ... 99.wav sequentially, in loop CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Accept the call (send 200 OK) immediately, play 100 files (the number 100 is configurable in script) sequentially from folder C:\WAV\. For every new call the counter is incremented and next audio file is played. Record the call (B side) for 20 seconds, end call leg B in 60 seconds, infinitely play recording CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Send DTMF digit some time after connection Script for user of multi-tenant softswitch. Note that you need to set field "CallXML variables" to "send_dtmf=true" for originators where you want to enable this feature. Blacklist caller RTP IP addresses that do not send RTP, to increase ACD of dialer traffic CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Overwrite B numbers and make sure that

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Number B is unique CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). The number lists can be uploaded via HTTP POST API method "UploadNumberList". Play ringback tone immediately when received a call, don't wait for call leg B (using "183 Session Progress" early media) CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Answer with "183 session progress" ("alerting") to call leg A after some random waiting time (1..3 seconds) and play a ringback tone from wav file in a loop. Also, make call to terminators according to routing configuration. Play ringback tone immediately when received a call, don't wait for call leg B (using "180 Ringing") CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Answer with "180 Ringing" to call leg A after some random waiting time (1..3 seconds). Then make call to terminators according to routing configuration. Request HLR to filter VoIP traffic (reduce SIM blocking) CallXML script for originator in StarTrinity softswitch (menu - configuration - originators - originator details - set "callxml script type" = "custom" - enter the script). Make API requests to hlrlookup.com: 1) validate caller number (A, CLI) 2) check status of destination number (B, CLD). Reject call with 503

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Opens a directory with log files. The files contain debugging information; by default error logs and GUI activity reports are sent to StarTrinity developers via encrypted channel. If you want to disable error reports, please set "DisableAnonymousReports" = "1" in settings tab. License information This screen displays information about license for current installation. For information about prices please contact sales manager by email. Command line interface (CLI) The SIP tester has a command line interface which is available with StarTrinity.SIPTester.CLI.exe. This executable runs in command prompt, it has following parameters: (if a parameter is not specified, it is taken from XML settings file) TotalCalls (required for active mode when generating calls) - total number of outgoing calls to generate, or 'infinite'. After exceeding this number, SIP Tester waits for current calls to complete TotalIncomingCalls (optional) - total number of calls to receive before exiting. After exceeding this number, SIP Tester waits for current calls to complete MaxCurrentCalls (optional) - max number of concurrent calls MaxCurrentCallsLimitMode (optional) - mode to limit max number of concurrent calls: 'incoming+outgoing' (default), 'outgoing', 'incoming' MaxTestDurationSec (optional) - max duration of test in seconds. If the "MaxTestDurationSec" is set, SIP Tester waits for all current calls to complete before exiting. Warning: please limit call duration by "maxansweredtime" to avoid hang calls IncomingCallsExitTimeoutSec (optional) - max time between incoming calls to stop test. Is used when you expect incoming calls to arrive during the test. If the "IncomingCallsExitTimeout" fires, SIP Tester waits for current calls to complete and exits with status -1 OutgoingCallsPeriodMs (optional) - interval between simulated calls in milliseconds NumberOfCallsPerBurst (optional, default is 1) - number of calls to generate per burst (tick) OutgoingCallsMode (optional) - mode of interval between simulated calls: fixed or random OutgoingCallXmlFile (optional) - CallXML file name to generate outgoing calls.. Download StarTrinity Softswitch latest version for Windows free. StarTrinity Softswitch latest update: Septem

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{ "@context": " "@type": "Event", "name": "Carriers World Asia, Bangkok", "startDate" : "2016-03-15T10:00", "url" : " "location" : { "@type" : "Place", "sameAs" : " "name" : "Carriers World Asia, Bangkok", "address" : "Centara Grand & Bangkok Convention Centre at CentralWorld, Bangkok" } } { "@context": " "@type": "Event", "name": "ITW 2016", "startDate" : "2016-05-08T10:00", "url" : " "location" : { "@type" : "Place", "sameAs" : " "name" : "ITW 2016", "address" : "HYATT REGENCY, Chicago, USA" } } { "@context": " "@type": "Event", "name": "CommunicAsia 2016", "startDate" : "2016-05-31T10:00", "url" : " "location" : { "@type" : "Place", "sameAs" : " "name" : "CommunicAsia 2016", "address" : "Marina Bay Sands, Singapore" } } --> --> An advanced VoIP tunneling software that helps VoIP service providers offer their services in geographical areas where VoIP calls are blocked by firewalls or bandwidth is low. Offered to iTel Mobile Dialer customers, iTel Byte Saver reduces bandwidth consumption up to 70% at the originating end. Using efficient tunneling technology, it passes calls to the VoIP service provider's Softswitch through any internet firewall or blockage by establishing a tunnel between itself and iTel Mobile Dialer. Furthermore, iTel Byte Saver ensures much less voice delay because of distributed media IPs. Key Features iTel Byte Saver can conveniently bypass firewalls through optimizing bandwidth in VoIP blocked and low badnwidth areas offering seamless VoIP calling experience for end users at all times. Passes calls through any internet firewall or blockage Distributed media for less voice delay Saves up to 70% on bandwidth consumption Compatible with any standard SIP Softswitch Handles 1000 concurrent calls per server Technical Specifications A server based tunneling software that uses various tunneling and data encryption methods to improve quality of VoIP communication while installed in between Mobile Dialer and Softswitch Supports G.729 codec Provides NAT support to prevent one way voice problem Controls RTP network wise packet size for smoother voice quality Data encryption for maximum security Use of UDP & TCP as transport method to bypass all packet filtering How it works iTel Byte Saver works efficiently in between the VoIP mobile client and service provider's softswitch. Calls travel from the mobile SIP client to iTel Byte Saver and by using its efficient tunneling technology, the software passes the call to the mobile VoIP service provider's softswitch. Further, the call gets registered and authenticated directly by service provider's SIP Softswitch. “iTel Byte

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AdoreInfotech.com, calling card, softswitch, voip billing, callback,call back, ippbx, pc2phone, ipphone2phone, wholesale voip solution,class4 switch, class5 switch, sip softphone, mobile dialer,CRM,Opensource solutionAdoreInfotech.com, calling card, softswitch, voip billing, callback,call back, ippbx, pc2phone, ipphone2phone, wholesale voip solution,class4 switch, class5 switch, sip softphone, mobile dialer,CRM,Opensource solutionWindows Mobile ApplicationWindows Mobile Voip DialerAdore Windows mobile Softphone will strategically add the power of live voice communication to any Smartphone to Phone and Smartphone to Smartphone. For Windows Mobile which supports SIP protocol especially designed to give you a smoothing communication experience and crafted out while keeping the various requirements of our clients. Adore Windows mobile for SIP protocol allows ITSP to customize the application with their logo and user interface. It can also be customize with local language to provide better service to local customer with their own native language. Adore windows mobile softphone use VoIP. It will allow customer to use mobile providers data plan to make international calls and saving up to 70% of their phone bill. This can use used in WIFI/GPRS/2G/3G networks. Adore windows mobile softphone can be used as SIP client for existing SIP IPBX for corporate users. It will allow user to make call from their mobile phone using their corporate phone system. This windows mobile dialer comes with the following set of features.Windows Mobile Calling Card DialerAdore Softphone has come up with “Callingcard Dialer” for Windows Mobile based Mobile phones. Windows Mobile Calling Card Dialer lets the callers Automatic dial the access number, Pin of prepaid card and Destination Number. The Windows Mobile Calling Card Dialer is meant for service providers who can use this Callingcard Dialer and get it integrated with their existing Callingcard server.. It works just like standard callingcard i.e. You may make a long distance phone call using a Callingcard or dial-through service .There is no need to remember your access numbers, Callingcard cards or PINs. Android ApplicationAndroid Voip DialerAdoreSoftphone has come up with " SIP Client for Android (Voip)" for android based Mobile phones. Android VOIP Mobile dialer lets the callers make VoIP calls through their standard Mobile phones. The Android Softphone is meant for service providers who can use this Softphone and get it integrated with their existing SIP servers. It works just like standard SIP app for Android i.e. it gets registered on the SIP server and then let the authentic caller makes calls. The biggest advantage that this Android voip provides is the huge base of potential customers who can now make calls using their standard mobile phones. The mobile phones need to be connected to the internet either through GPRS/EDGE/Wi-Fi/3G.Android Calling Card DialerAdore Softphone has come up with “Callingcard Dialer” for Android based Mobile phones. Android Calling Card Dialer lets

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IP address of initial SIP INVITE packet Call-ID - unique identifier of SIP call, "Call-ID" header in SIP messages. Clicking on this column shows trace of SIP messages Disconnection status code - indicates reason of destroying a call StarTrinity SIP Tester introduces a set of custom codes to indicate abnormal call termination: 1408 (NoResponse) - receiving no response from destination to initial INVITE (request timeout) 1409 (MaxConnectedTime) - call was terminated because of configured "max connected time" in CallXML script 1410 (SessionTimerExpired) - RFC4028 timer expired 1411 (CallMaxLifetime) - call was aborted because of global setting "MaxCallLifeTimeInHours" 1580 (IdleMediaDetected) - call was aborted because of "idleMedia" event, when no RTP was received from remote side during "IdleMediaTimeToEndSession" seconds (see global settings) . State - state of INVITE session: Null - before INVITE is sent or received, e.g. when resolving host name or establishing TCP connection Calling - after INVITE is sent Incoming - after INVITE is received Early - after response with To tag Connecting - after 2xx is sent/received Confirmed - after ACK is sent/received Disconnected - when terminated Released By - identifies party which ended a call: "caller" or "called" Answer delay - time between sending INVITE request and receiving 200 OK response 100 delay - time between sending INVITE request and receiving 100 (Trying) response 180 delay - time between sending INVITE request and receiving 180 (Ringing) response 183 delay - time between sending INVITE request and receiving 183 (Session Progress) response 180/183 delay - time between sending INVITE request and receiving 180 (Ringing) or 183 (Session Progress) response. Can be used to measure signaling-based post-dial delay (PDD) Audio signal delay - time between sending/receiving INVITE request and receiving RTP audio signal which is greater than -24dB (default value). In active mode SIP Tester analyses RTP audio

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Sample CallXML Scripts for StarTrinity SIP Tester The scripts should be copied from this page into SIP Tester -> Outgoing calls simulation -> XML tab, into root element, replacing previous code. Generation of random outgoing calls This is a script for stress testing call centers, PBX systems, outbound dialers with SIP Tester. It generates calls with specified prefix, random caller ID (A number) and called ID (B number). Destination SIP URI is localhost:5061, it has to be replaced with actual URI. Registering range of IP PBX extensions and making calls between them Reading IP PBX extensions, registering and making calls between them Specifying custom SIP headers and SDP attributes You can add custom SIP headers by setting attribute "headers", also add SDP attributes by setting attribute "sdpAttributes". This could be helpful for testing ED-137 SIP applications. headers="Subject=radio|Priority=normal|WG67-Version=radio.01" sdpAttributes="R2S-KeepAlivePeriod:200|R2S-KeepAliveMultiplier:10|sigtime:1" /> Test IVR menu 1 The script reads caller ID, called ID and DTMF sequence from CSV file, creates a call and sends DTMF digits to destination. Test IVR menu 2 Another script example simulates several DTMF keys and records audio into WAV file. Recording contains both debugging prompts with synthesised speech and audio from tested IVR. Test IVR menu 3 (dead air + recording) The script makes call to a number, waits for IVR playback. If there is silence (dead air), it sends an alert email (settings MailSender***). Minimal interval between emails is 5 minutes (sending emails is limited to avoid too many emails). All calls are recorded to WAV files.

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Header is generated from CallXML session ID. Via: SIP/2.0/UDP 192.168.206.197:5060;branch=z9hG4bK18171d2b3f11 From: "2000" ;tag=104574~6a5c17ac-9109-44e8-b315-975d70cd1b5d-22861245 To: Date: Mon, 29 Jul 2013 20:55:20 GMT Call-ID: SipTester$Id; Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM8.6 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence Send-Info: conference, x-cisco-conference Alert-Info: Contact: Remote-Party-ID: "2000" ;party=calling;screen=yes;privacy=off Max-Forwards: 69 ]]> Sending custom SIP INVITE with 2 media streams in SDP The script sends custom SIP message to server without creating call (only for testing SIP stack). The message is specified by 'CDATA' content of XML element. SIP Tester automatically calculates 'Content-Length' header, so only 'Content-Type' header is needed. Call-ID header is generated from CallXML session ID. Via: SIP/2.0/UDP 192.168.206.197:5060;branch=z9hG4bK18171d2b3f11 From: "2000" ;tag=104574~6a5c17ac-9109-44e8-b315-975d70cd1b5d-22861245 To: Call-ID: SipTester$Id; Supported: timer,resource-priority,replaces User-Agent: StarTrinity SIP Tester Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Contact: Max-Forwards: 69 Content-Type: application/sdp v=0 o=root 342983111 342983111 IN IP4 192.168.1.188 s=Asterisk PBX 1.6.2.15 c=IN IP4 192.168.1.188 t=0 0 m=audio 11568 RTP/AVP 18 8 0 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv m=image 4488 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:397 a=T38FaxUdpEC:t38UDPRedundancy ]]> Monitoring SIP server with email notifications This script makes a call to a SIP server and sends email alarms on success and failure. The SIP Tester Tool could be configured execute script periodically. Note that interval between calls has to be large enough to avoid anti-spam protection at mail server. The sendemail element uses settings "MailSenderUserName", "MailSenderPassword",. Download StarTrinity Softswitch latest version for Windows free. StarTrinity Softswitch latest update: Septem

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Lawful Interception features are built-in to the PBX application. Softswitch : The CommuniGate Pro server-based software solution combines the functionality of class 4 and class 5 systems into a single platform, delivering end-to-end connectivity and PBX voice and conferencing applications to all subscribers. It provides Class 4 services with SIP signalling and call control or replacement of Class 4 TDM systems and allows Class 5 services with PBX, voicemail, audio conferencing, and auto-attendant applications or replacement of Class 5 TDM systems. Conference Server : A robust and easy to use voice conferencing server allows you to host thousands of simultaneous .virtual conference. bridge calls, available to both PSTN and VoIP callers. Running your conference server in-house will greatly improve your call quality and ease of use while decreasing costs spent on conference service outsourcing. Call recording and archiving enables the ability to capture and store meeting information. Auto-Attendant Application : Default support for simple office scenarios and extensible through the CG/PL language to fit any needs, including multiple language support. Can be extended for full Interactive Voice Response (IVR) and Call Center services such as Automatic Call Distribution (ACD) and caller queuing. Voicemail Application : Developed for typical office needs, and is fully extensible for carriers that need to provide voicemail for millions of subscribers. Groupware Integration : CommuniGate Pro Real-Time Applications are not stand-alone point solutions, but are integrated into the email and groupware calendar unlike any communications platform on the market today. CommuniGate Pro provides click-to-call from the