RE: Can't understand how transfer calls from digital recept. to classic ext., pstn
Please provide us with more information regarding this issue.
• How are you calling the Digital Receptionist (DR)?
---> From within PBXes or from a PSTN DID via an inbound Trunk?
• Can you set the Inbound Route to bypass the DR and dial directly the Classic extension?
---> Will the inbound call ring on the Classic extension?
• Is the trunk used to dial the PSTN number of the Classic extension, the same trunk the inbound call comes in?
---> Will the ITSP allow concurrent calls on this trunk, or just allow a single voice channel?
RE: Can't understand how transfer calls from digital recept. to classic ext., pstn
1. From DID via inbound.
2. No. If bypass DR - problem the same, buzy tone after few sec. In logs I have:
Nov 19 11:20:42 VERBOSE[89460] logger.c: -- Called 201117878899@from-internal/n
Nov 19 11:20:42 NOTICE[89472] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Nov 19 11:20:42 VERBOSE[89472] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Nov 19 11:20:42 VERBOSE[89460] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Nov 19 11:20:42 VERBOSE[89460] chan_sip.c: Hangup call SIP/0290636271-fb59, SIP callid 7cdca30f499f45e85bf2eec31e14fc6e@80.85.244.180
3. I don't understand at all how to manage wich trunk will be used when call go from inbound to pstn... with or without DR. Concurrent calls accepted on inbound calls trunk.
