Sip ip phone adapter 52 ports asterisk sip trunk voip gateway for call center

     Sip ip phone adapter 52 ports asterisk sip trunk voip gateway for call center

52E1 VoIP Media Gateway with SIP trunk NC-MG1000, support SS7, ISDN PRI, V5.2, R2, CAS, Q.SIG, DPNSS to SIP

Compitable with Asterisk, Cisco call manager, Free switch, 3CX, Mircrosoft Lync, VOS, etc.

Configure, Monitor, Management, Backup, Update: by RS232 console port, Telnet, WEB.

Our service: 7*24 supports for ever.

Warrantly: three years.

Delivery time: 1 day.

Advantage:
1. Modular design with up to 8 E1/T1 ports (Two modules, Each module supports maxima 4E1);
2. Complete Signaling (SS7/ISDN PRI/V5.2/Q.SIG/R2/SIP/T.38);
3. The best device for Call Center;

NC-MG1000 is a cost effective and standards-complaint VoIP Media Trunk Gateway, which is generally called TG/SG or VoIP Voice Gateway, designed for enterprise users and ITSP (Internet Telephony Service Provider) environments.

NC-MG1000 locates on the interface of PSTN and IP network, providing signaling transform between SS7, ISDN PRI, V5.2 signaling and SIP Protocol, as well as media transfer between the PSTN and IP network.

Features:

·Echo Cancellation: G.168 128 ms Network Echo Canceller;
·Fax Support T38 and pass through;
·64 conference call support;
·Generic Voice Activity Detection (VAD);
·Generic Comfort Noise Generation (CNG);
·Calling control: Called/Calling party number translation, second stage dialing, Auto dialing with DTMF, Ring Back tone;
·Flexible call routing management;

Capacity:
·A 1U equipment can have 2 independent VoIP modules;
·Each Module contains:
-1~4 E1;
-128 G.711 Channels;
-64 G.723, G.726, G.729 Channels;
-32 T.38 Fax Channels, 64 Conference Bridge;

Signalings, Protocol and Codec that support by NC-MG1000:

·Signaling support:
-SS7 (ITU-T Q.700 series), 24 bits/14 bits PC, ISUP/TUP;
-ISDN-PRI (ITU-T Q.931, Q.921);
-V5.2 (ITU-T G.964, G.965)
-R2/CAS

-Q.sig

-DPNSS
·VoIP protocol support:
-SIP
 RFC3326(Reason header in SIP messages)
 RFC3372(SIGTRAN and SIP-T)
 RFC2327(sdp)
 RFC3398(ISUP-SIP Mapping)
 RFC3261(sip)
 RFC5806(Diversion Indication in SIP)
 RFC2833( DTMF )
 RFC3362(t.38)
 RFC 3261(SIP 2.0)
 RFC3204(MIME media types for ISUP and QSIG Objects)
 RFC3578(Mapping of ISUP overlap to SIP))
-Voice Codec:G.711 U-Law and A-Law, G.711 Appendix 1, G.723.1 and G.723.1 Annex A, G.729 Annex A and Annex B, G.726,GSM,ARM,ILBC;