3 Way Calling
Accidental Disconnect Protection
Answer Groups/Keys
API -
• C
• Perl
• Python
• XML Via Perl
• Shell scripting
Autodial Keys 7960: yes
Automated Attendant -
• Custom Call Routing
• System Answer
Automatic Line Selection
Automatic Set Relocation Assumes use of DHCP
Auto Attendant
Auxiliary ringing Use ATA-186 or other adapter to connect to additional analog ringers
Background Music mp3 streaming or files standard. Line-in audio level 1 effort
Busy Lamp Indication NA Phones can only display local number status
Button Inquiry
Call Detail Record ASCII, MySQL, Postgres, ODBC
Call Display when busy Shows number of inbound call-wait callers on screen of 7960
Call Duration timer Built into 7960
Call Generation Creation of calls by API or script activation (e.g.: click a mouse, two phones ring and are connected)
Call Forward-Fixed -
• All calls
• Call Forward Busy
• No Answer
• Override
Call Forward -
• All calls
• Busy
• No Answer
• Override
• Variable
Call Park
Call Pickup
• Directed
• Pickup Groups
• Remote
• Trunk Answer Any Station
Call recording For call centers or “for quality assurance…”
Call Queuing
Call Waiting
Camp On 1
Central Answering Position
• CAP feature & autodial only
• BLF indication
Class of Service
• COS Passwords
• Dialing Abilities
CMS/CLASS -
• Call Information Session
• Caller Log
• Calling Name Display
• Calling Number Display
• LOGIT Feature 1 LOGIT is a manual log feature
• Long Distance Indicator
CODECS -
• G.711U
• G. 711A
• G.729a With additional one-time per channel cost
• G.729b Rarely used; “a” usually all that is required
• ILBC
• GSM Note: not compatible with Cisco GSM format
• G.723A NA Supported as passthrough, but not transcoded
• G.723B NA Supported as passthrough, but not transcoded
• ADPCM
Conference
• Meet Me Can bridge all lines into “conference room” with selectable mute
Delayed Ring Transfer
DID Digit Translation
Dialing Modes -
• Standard
• Automatic
• Predial
Dial Mode for Lines - pulse/tone
Dial Pad Feedback
Direct Station Select Buttons NA
Disconnect Supervision Analog tones less efficient than Q.931 PRI signalling
Discriminating Ringing at Set NA
Distinctive Ringing Cadence S Each deskset can have a different ring tone (customizable, uploadable)
Do Not Disturb
Enhanced Call Restrictions
External Line Access Code
Flexible Numbering Plan
Group Listening S See “meetme” description
Handsfree
Hold
• Held Line Reminder Visible indicators, but no audible
Hot Line-Ringdowns ATA-186, Grandstream: yes
Hunt Groups -
• Circular
• FIFO S
• Sequential
Incoming Line Groups
Interactive Voice Response DTMF only; no voice recognition without additional work
Language Choice Yes; phones can have alternate language sets, and system can use multiple voice prompts with small work
Last Number Redial
Line Button Relocation Six choices on 7960; can move lines between any of those buttons
Line Names
Line Pool(s)
Line Redirection
Link/Flash (Recall)
Listen On Hold AKA Mute
Long Tones DTMF duration can be set system-wide
Multiple Line Appearances
Music/Tone/Silence On Hold
Networking - Norstar to M-1 NA
Night Service Automatic easily done; manual level 1 effort
Onhook Dialing -
• Automatic Dial
• Predial
• Standard
Paging -
• Internal (Multiple Zones) 7960, Polycomm Soundpoint: yes
• Page Yes/No Per Set 7960, Polycomm Soundpoint: yes
• Page time-out NA
• Page tone on/off 7960, Polycomm Soundpoint: yes
Preselection/Call Screening Filters based on CID
Prime Line Select
Prime Set(s)
• Multiple prime sets
Priority Call Can be made to bump calls from PRI groups based on priority of new calls
Privacy -
• On/Off Requires DTMF sequence or with slight amount of work, an XML interface
• Per Line
PSTN Gateway Hardware -
• Single FXO port (PCI card)
• Quad FXO port (PCI card) Voicetronix: now, Digium: Q2 2004
• Single T1/E1 port (PCI card)
• Quad T1/E1 ports (PCI card)
Ringing Line Preference This means call queues, with the longest ringing line appearing first in order
Remote System Access Remote admin via standard UNIX tools (ssh, etc.)
Receiver Volume
Remote Users -
• System supports distant users Any TCP/IP network can provide transport for voice data
• UA devices work behind NAT Works without STUNd or hardware agent; self-contained NAT traversal
Restriction Override Password(s)
Ring Again (Internal) If a call is made to internal line, and no answer, then ring back when phone next used
Routing Service/Dest. Codes
Saved Number Redial
Service Modes Described as enhanced business hours (night mode) service
SIP call handling
Speed Dial: Personal Easily added, but web interface takes more effort
Speed Dial: System
Time /Date Display
TOS/QoS IP header bit setting
Transfer -
• Immediate/Blind
• With announcement
• With Callback
• Over Public Network
Unified Messaging -
• email delivery of voicemail
• web delivery of voicemail Again, assumes a larger authentication/web infrastructure
• pager notification of email
Unsupervised Conference
User programmable feature keys NA
Voice Call
Voice Call Deny
Voicemail Storage -
• selectable encoding formats
• assignable limits (space, time)
• voicemail file transfer via ftp/nfs/etc.
No tag for this post.
Salaam Syed,
Can Asterisk provide answer supvervision based on timer.
e.g. When a call comes into the Asterisk, we program it to say provide a tone say after 35 secs to say the call is answered.
Thank You