Dialogic IMG 2020网关

来自Asterisk Freepbx官方最权威最新中文技术文档资料,分享呼叫中心配置资料-asterisk,freepbx,Issabel 用户手册 界面配置,呼叫路由,IVR, 网关对接,拨号规则,SIP 分机呼叫,pjsip, IVR, 录音, CDR, 队列呼叫,振铃组,CLI 命令中文资料手册
跳转至: 导航搜索

概述:

  • Dialogic ® IMG 2020集成媒体网关通过高密度光,电话和以太网在一个紧凑的1U外形的连接能够进行IP和PSTN网络之间的互通。它将媒体和信令转换为支持基于移动和基于云应用的高效可靠的语音,传真,调制解调器和基于音调的会话。IMG 2020通过IP和混合网络边界保护会话,以支持服务级别协议。
  • IMG 2020内的单个机箱中的时分复用(TDM)与IP网关,任意至任意信令和会话边界控制器(SBC)功能的组合提供了与较少的CAPEX和OPEX相比显着降低的潜力综合替代品。
  • 除了提供广泛的可扩展性外,IMG 2020还可以在单一运营商级机箱中处理信令和媒体,并可将SIP服务提供到SS7,SIGTRAN,PRI和SIP-I网络。包括支持IPV6,嵌入式防火墙,拒绝服务(DoS)保护和批量SIP注册等选定SBC功能的整合,便于客户从TDM固定和移动网络迁移到IP网络。这使客户能够更新其网关以支持新的服务,例如高清(HD)语音,IP网络和SIP中继之间的转码。


Dialogic IMG 2020.png
Dialogic 2020 Integrated Media Gateway

主要特征:

  • 可支持50到2250个同步SIP会话与多媒体转码,以及128到2016个通道的七号信令
  • 在单一平台上结合了IP和TDM网关功能
  • 运营商级设计和功能提供高可用性,可靠的吞吐量和增强的服务交付
  • 无需添加单独的硬件来支持安全性和转码需求,有助于降低CAPEX和部署的平台数量
  • 支持任何信号和媒体
  • 支持SS7,SIP信令以及IPv6和IPv4互通以及语音转码提供了一个经济高效的平台,可帮助服务提供商从TDM发展为全IP环境
  • 可扩展的IP和TDM连接解决方​​案以小尺寸提供高性能,有助于降低OPEX和CAPEX
  • SIP分析器,基于Web的用户界面和离线配置
  • 集成的加密和转码支持语音,语音和传真
  • 集成的多媒体网关功能有助于TDM和IP互通,以提供服务交付灵活性和域之间的自动故障转移
  • 易于使用的服务配置和管理工具可帮助加速服务部署并简化平台管理

功能参数介绍:

  • 呼通功能:

Calling Party Category、Generic Name Parameter、ISUB Encoding、ISUP Cause Code Mapping、Map SS7 ISUP Messages to SIP、Message Waiting Indicator、Overlap Signaling (Sending)、Overlap Signaling (Receiving)、Originating Line Info Support、Parameter Mapping、P-Charge Header、Reason Header - Local/Remote Resource Availability、SIP P-Charging-Vector Header、SIP CIC and DAI Codes、SIP to SIPT Overview、Signaling、Suppress Local CPT when CPG Indicators Received、TCAP CNAM Database Queries、Translate SI and PI from SIP to SS7、UUI Support、Multi-Level Precedence and Preemption - MLPP、SIP Header Indicates Multimedia Call、Map ISDN Call Proceeding to SIP 183、SIP Precondition and SS7 Continuity Check Protocol Interworking、SIP P-Early-Media and SS7 UID Parameters Interworking、Interwork Connected Number between SIP and ISUP、Interwork Connected Number between SIP and ISDN

  • H.323功能:

H.323 Overview、H.323 Supported Messages and Protocols、H.323 Fast Start、H.323 Fax/Modem Support、H.323 Keep Alive Timer、Incomplete Call Behavior、H.245 Tunneling

  • ISDN功能:

ISDN Overview、Two B-Channel Transfer (TBCT)、Explicit Call Transfer (ECT)、Release Link Trunk (RLT)

  • 路由功能:

Routing Overview、Digit Matching、Digit Translations、Wildcards for Routing、Wildcards for Translation、Generic Number Translation、Digits 0x0A to 0x0F Support、Multimedia Call Routing、Route List Percentage Based Routing、Local Number Portability (LNP)、Route Call to a Cause Code、Graceful Out of Service、Time of Day Tables、Time of Day Table、Time of Day Element

  • Cause Codes:

Cause Code Values、Cause Code Mappings、Cause Code Conversions、Reject Call with a Cause Code、Customize / Map Cause Codes

  • SIP功能:

SIP Overview、SIP Load Balancing、SIP Delayed Media Inbound - 3PCC、SIP Delayed Media Outbound、SIP Call Hold、SIP Session Timer Overview、SIP Session Timer Call Flows、Pass Through + Character in the、User Part of URI、SIP FQDN Support、SIP Proxy - Overview / Configuration、SIP Retry-After Header - Transmit、SIP Retry-After Header - Receive、SIP Multiple m-line Support、SIP Enum Support、Multiple、SIP Addresses - Overview、Multiple SIP 183s prior to 200 OK、Delayed Answer、DNS Query using SRV Record、SIP Options - Busy Out/Keep Alive Configure、SIP Redirect、SIP Redirect - Initiated 302 Response、SIP Diversion Header、Diversion and RPID Interworking SIP to SS7 REL、SIP History-Info Header、SIP Reason Header、Privacy、SIP Phone Context Parameter、SIP INFO Method - DTMF、SIP INFO METHOD - Subscribe/Notify、SIP INFO Method - Long Call Duration Auditing、SIP INVITE Method、SIP Update Method、SIP REFER - Call Transfer、SIP Codec Negotiation、SIP Fax / Modem Support、SIP Overload、Control Overview、SIP PRACK、SIP PRACK Call Flows、I20-Signaling、Initiate 3xx Response to Endpoint、Propagate From Header (SIP to SIP)、SIP Message and Header Restrictions、tel URI、Passing P-Asserted-ID with two URIs (SIP to SIP)、Passing Referred-By Header (SIP to SIP)、Passing Unsupported SIP Headers (SIP to SIP)、Precondition and P-Early-Media Header Support、Call Answer Timeout and 183 Periodic Retransmission functionality

  • SS7功能:

SS7 Overview、SS7 Link OOS Management、Sigtran / M3UA、SS7 Redundancy、ISUP Explicit Call Transfer

  • VoIP功能:

AMR-WB、Clear Channel Codec Support、Ethernet Redundancy via Smart Probe、Media Inactivity Timers (RTP)、RTP Source Port Validate、Supported Codecs、Symmetric NAT Traversal、Transcoding、VLAN Tagging、Vocoder Order - Rearrange、SIP Media Flow Attribute Support

  • 安全:

IP Security、Denial of Service (DoS/DDoS)、Provisional ACL、Secure HTTPS/SSH、SIP over TLS、SRTP

  • 视频转发单元:

Video Fast Update

  • 呼叫跟踪:

Calling Party Information Retrieval via INR/INF、Call Trace CUG passed、Call_Trace_CUG_rejected、Call Trace - 180 Ringing (F-6480)、Call Trace - 183 Session Progress (F-6480)、Call Trace - Initiate 3xx、Call Trace ISUB ENC ISDN to SIP、Call Trace ISUB ENC SIP to ISDN、Call Trace LNP (SIP-SIPI)、Call Trace MWI、Call Trace OLI ISDN-SIP、Call Trace OLI SIP-ISDN、Call Trace - P-Charging-Vector header、Call Trace CNAM - SS7 to SIP、Call Trace - ENUM、SIP INFO Subscribe-Notify Call Trace、SIP Phone Context Call Trace、Call Trace Spirou SIP to SS7、Call Trace Spirou SS7 to Spirou、RADIUS - Pre-Paid Call Trace、RADIUS - Pre-Paid Routing Call Trace、SIP to SS7 Call Trace - Overlap、SRTP - Inbound 183 Session Progress、SRTP - Inbound 200 OK、SRTP - Inbound INVITE、SRTP - Outbound 183 Session Progress、SRTP - Outbound 200 OK、SRTP - Outbound INVITE、SS7 to SIP Call Trace - Overlap、UUI Call Flows - Binary、UUI Call Flows - Hex

注:具体详细的配置请查看Sangoma官方wiki:https://wiki.freepbx.org/display/DIMG/IMG2020