Progressinband Settings

  • Progressinband:
    • yes – when the "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio.
    • no – send 180 Ringing if 183 has not yet been sent, establishing an audio path. If the audio path is established already (with 183), then send in-band ringing (this is the way Asterisk historically behaved because of buggy phones like Polycom's).
    • never – whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behavior of Asterisk.

NOTE: if Progressinband does not work, add "prematuremedia=no" to sip.conf and reload Asterisk.

  • Block callerid if (number) simultaneous calls come from it – blocks CallerID if the entered number of simultaneous calls come from it
  • Limit up to (number) calls, during (number) seconds – allows to set calls per second limit in some period. File:call_limit_per_period.png
  • Outbound Proxy – send outbound signaling to this proxy, not directly to the peer (Internal Asterisk option).
  • SIP Session Timers - SIP Session Timers provide an end-to-end keep-alive mechanism for active SIP sessions. Possible values are "accept", "originate", "refuse":
    • originate - request and run session-timers always.
    • accept - run session-timers only when requested by other UA.
    • refuse - do not run session timers in any case.
  • SIP Session Refresher - The session refresher (uac|uas). Defaults to 'uas'.
    • uac - default to the caller initially refreshing when possible.
    • uas - default to the callee initially refreshing when possible.
  • SIP Session Expires - maximum session refresh interval in seconds. Defaults to 1800 secs.
  • SIP Min Session - minimum session refresh interval in seconds. Defaults to 90 secs.
  • Hangup Call if PDD is more than - set maximum PDD time (in seconds) before call ir hangup. Note that failover providers will not be used in this case. Call will be hanged up completely and will not be passed to other provider in LCR.
  • Localize PAI - if set to 'yes' then MOR will localize originator's number from PAI (P-Asserted-Identity) by CallerID Localization Rules. Works only if pass_pai is set in mor.conf.
  • Pass PAI - available options are 'Global', 'Yes', 'No'. Global (default value) means use value from pass_pai in mor.conf file. More information available in table here.
  • Use random Number when CallerID is invalid - if set to Yes then when CallerID is NOT found in a Number Pool set in Number Pool with valid CallerIDs it is changed to a random CallerID from Number Pool set in Random Number from a Number Pool.
  • Emergency CallerID - number from this field will be used as CallerID if Destination number matches any number from Emergency CallerID Number Pool (see below). Localization rules will be applied before checking Destination number in Number Pool. P-Asserted and/or Remote-party-ID will be overwritten with the same as CallerID. If Destination number (after localization applied) does not match any number in Emergency CallerID Number Pool, then CallerID is handled as usual.
  • Emergency CallerID Number Pool - Number Pool with numbers used for Emergency CallerID feature.
  • Music On Hold to Device - what to play when this Device is made On Hold by other party. Feature named "mohinterpret" on Asterisk.
  • Music On Hold from Device - what to play to other party, when this Device sets it On Hold. Feature named "mohsuggest" on Asterisk.
  • Announcement to the Called Party - plays specified announcement to callee in the beginning of conversation, right after call is answered. For example you can play message like "This conversation will be recorded" before every call from this Device.

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