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Thread: RE: Can't understand how transfer calls from digital recept. to classic ext., pstn
ers

Replies: 2
Views: 10160

RE: Can't understand how transfer calls from digital recept. to classic ext., pstn 19.11.2011 11:31 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

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.

Thread: RE: Can't understand how transfer calls from digital recept. to classic ext., pstn
ers

Replies: 2
Views: 10160

Can't understand how transfer calls from digital recept. to classic ext., pstn 30.07.2011 14:47 Forum: Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups

All sip extensions from dig. rec. works ok, classic ext. when I call it directly also ok, but classic ext. from dig. rec. not working.

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