Difference between revisions of "MOR Features"
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| SIP || [[Image:icon_check.png]]|| max packet size 4096 bytes | | SIP || [[Image:icon_check.png]]|| max packet size 4096 bytes | ||
|- | |- | ||
| H323 || [[Image:icon_check.png]]|| | | H323 || [[Image:icon_check.png]]|| disabled by default because its instability | ||
|- | |- | ||
| IAX2 || [[Image:icon_check.png]]|| | | IAX2 || [[Image:icon_check.png]]|| disabled by default because its instability | ||
|- | |- | ||
| E1/T1 (PRI), BRI, FXO/FXS || [[Image:icon_check.png]]|| With additional hardware cards | | E1/T1 (PRI), BRI, FXO/FXS || [[Image:icon_check.png]]|| With additional hardware cards | ||
Revision as of 15:30, 5 January 2017
This page is not complete yet. Not all Features are here.
Protocols
| Feature | Support | Comments |
| SIP |  |
max packet size 4096 bytes |
| H323 | |
disabled by default because its instability |
| IAX2 | |
disabled by default because its instability |
| E1/T1 (PRI), BRI, FXO/FXS | |
With additional hardware cards |
| SRTP | |
|
| SIP TLS | |
|
| H.323 Fast Start | |
|
| H.245 Tunneling | |
|
| H.248 |  |
|
| SS7 | |
|
| XMPP | |
|
| ENUM | |
|
| PRACK | |
|
| SMPP | |
Codec Support (Audio/Video Compression)
| Feature | Support | Comments |
| Codec Auto-Negotiation | |
|
| Codec Transcoding | |
|
| RTP Multiplexing | |
|
| G.711 A-law | |
|
| G.711 u-law | |
|
| G.723.1 | |
|
| G.726 | |
|
| G.729A | |
|
| G.729, G.729B, G.729AB, G.729F, G.729D | |
More info |
| GSM | |
|
| iLBC | |
|
| LPC10 | |
|
| Speex | |
|
| ADPCM | |
|
| 16 bit Signed Linear PCM | |
|
| H.261 Video | |
|
| H.263 Video | |
|
| H.263+ Video | |
|
| H.264 Video | |
|
| G.722 | |
Supported from MOR 11 |
These codecs are included (not only supported) in MOR.
PBX Features
| Feature | Support | Comments |
| Call Logging | |
|
| Call Reporting | |
|
| Blind Call Transfer | |
Consult your phone manual |
| Attended Call Transfer | |
Consult your phone manual |
| Call Forward on Busy | |
Implemented with Call Flow |
| Call Forward on No Answer | |
Implemented with Call Flow |
| Call Transfers/Forward on Phone | |
By default are not available, but it is possible to enable this functionality. More information why transfers are not supported well in Asterisk can be found here |
| Call Routing (DID) | |
|
| Caller ID | |
|
| Conference Calling | |
Possible workaround described here |
| Conference Rooms |  |
Possible over external server |
| Auto Attendant / Digital Receptionist / IVR | |
|
| Voicemail | |
|
| Music on Hold | |
Using default Asterisk MOH |
| Ring Groups | |
Supported from MOR 11 |
| Hunt Groups | |
|
| Central Phonebook | |
|
| Call Parking | |
|
| Call Pickup | |
|
| Call Queuing | |
Possible using External DID functionality |
| Call Recording | |
|
| Dial by Name | |
|
| MWI - Message Waiting Indicator | |
Supported from MOR 9 |
| BLF Status Updates | |
Partly support |
| Intercom | |
|
| Paging | |
|
| Automatic Ring Back | |
|
| Skype interconnection | |
|
| Follow me | |
Implemented with Call Flow |
| Gatekeeper | |
|
| FAX passthrough | |
|
| FAX2Email | |
|
| SIP Re-Invite | |
|
| Rport extension RFC3581 | |
DTMF
| Feature | Support | Comments |
| RFC2833 | |
|
| inband | |
if system includes SIP Proxy, only RFC2833 is supported |
| info | |
if system includes SIP Proxy, only RFC2833 is supported |
Management and Scalability
| Feature | Support | Comments |
| Web-based management console | |
|
| Configuration Wizard | |
|
| Real Time Web-based System Status | |
|
| Integrated Web Server | |
|
| Automated Restore and Backup | |
|
| Firewall Friendly Configuration of External Extensions via Tunnel | |
|
| MS Windows Server Certified | |
No need, because it is built on Linux |
| Integrated Enterprise Database (MySQL) | |
|
| Run as Virtual Machine | |
|
| NAT friendly tunnel feature | |
|
| Remotely manage IP phones | |
|
| Automatic Phone Provisioning | |
|
| Allow Users to Configure Own Extensions | |
|
| Multi-Level Resellers | |
Workaround will be available later |
