I saw that mentioned in another thread and tried it, to no avail. However, obviously, this has the side-effect of disabling call waiting globally.įor what it's worth, setting ptt.* options such as to 0/disabled does not resolve the issue. If I set ="0" in addition to the above, this causes the incoming intercom call to ring until the PBX sends the call to voicemail, and the called phone won't answer it if already on another call. However, this is a half-baked workaround and not a solution, and wouldn't fit the use-cases of many or even most users, I would imagine. This will return a busy signal to the calling phone and the caller hears a slow-busy. One of the ways I have found to fix this is to set call.callsPerLineKey to 1.
#Freeswitch autoanswer registration#
For obvious reasons this is problematic.Īs mentioned, the config I'm running is barebones/default for the most part, other than the registration configuration and auto-answer configuration, so this is apparently default behavior in 5.2.5. except that it also auto-answers calls even if the phone is already on a different active call, and places the previous call on hold. This sets it up so that the polycom auto-answers calls that have the "intercom" indication in the SIP headers. So it's using the default settings for the se.rt.autoAnswer class. Other than the line registration and outbound proxy settings, everything else is default, from a freshly factory-reset VVX 600. Intercom is enabled with the following config lines:
#Freeswitch autoanswer software#
Otherwise, everything is the same as in caseįull text log of the session: 7.I can confirm that the auto-answer interrupt is occurring on UC software 5.2.5, on a VVX 410 and VVX600. Is pressed in the ASTPP GUI, a simulation of the termination of the call by
The procedures are the same as in case one.īased on the logs, when the HANGUP button Information is entered into the database, unlike case five.Ī calls ( invalid number with special characters )Ĭall information is not entered into the database, unlike case four. When instructions from XML are executed byĭialplan, an "origination rates not found" error occurs. The ORIGNATION_RATE_NOT_FOUND line is entered into the two last fields: The session ends after playing A voice auto answer.Ī calls 688444 (number non-existent in DB)īut an incorrect request to the database occurs, in the generated XML document, Case three: A calls B, B doesn'tīut after 1m In the RINGING state Freeswitch ends session with USER_BUSY error. The procedures are the same as in case one,Īfter callstate becomes RINGING for A and B, B terminated session:ĬALL_REJECTED procedure. Has been interrupted, start HANGUP procedure and change callstate to HANGUP forįreeswitch changes state to DESTROY and begins closing session (342-364 s trings in the log):įull text log of the session: 2.txt Case two: A calls B, B decline The phone and change callstate to ACTIVE for A:Ī and B exchange audio (294-308 strings inįreeswitch performs HANGUP hangup procedureĪnd change callstate to HANGUP for B ( 304-324 s trings in the log). The state one by one for B side, sinceĮverything is fine with the data and it is ready to continue working:Īudio codecs (51-55 strings in the log) and change callstate to ACTIVE for B:įreeswitch informs A that B has picked up Mod_dptools now also executes instructionsįrom an XML document (165-202 strings in the log):įreeswitch sets proxy route 40143 (B account) and sending invite:įreeswitch performs the routing process (B account number) (231- 244 strings in the logs) and enters XML document (129-158 strings in the log): More about mysql database usage by ASTPP: Ĭall process necessary information (90-128 strings in the log):įreeswitch performs the routing process forĪ (51-55 strings in the log) and puts the call for A in the RINGING state:ĭiaplan now executes instructions from an Upcoming session for 8620826254 (A number):įreeswitch sends SDP message ( further in the document will not focus on SDP messages):Īudio codecs for A (27-43 strings in the log)): In this example, two accounts are assigned: Must be followed from the beginning till the end: Call Packet Tracking (fs_cli) Case one: A calls B, B answers, B Six: A calls, B answers, hang up from ASTPP GUI (Switch => Live calls).