PBXes » English » Queues, Digital Receptionist, Faxmail, Voicemail and Ring Groups » RE: Can't understand how transfer calls from digital recept. to classic ext., pstn
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ers


Registration Date: 01.01.1970
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Can't understand how transfer calls from digital recept. to classic ext., pstn Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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.

30.07.2011 13:47 ersn is offline Search for Posts by ersn Add ersn to your Buddy List
Dia
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Registration Date: 03.03.2006
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RE: Can't understand how transfer calls from digital recept. to classic ext., pstn Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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?

19.08.2011 11:32 Diafora is offline Search for Posts by Diafora Add Diafora to your Buddy List
ers


Registration Date: 01.01.1970
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RE: Can't understand how transfer calls from digital recept. to classic ext., pstn Post Reply with Quote Edit/Delete Post Report Post to a Moderator       IP Information Go to the top of this page

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.

19.11.2011 10:31 ersn is offline Search for Posts by ersn Add ersn to your Buddy List
 
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