Re: [asterisk-users] asterisk and norstar
Thanks for the response!I've been reading and trying things and I cannot find a way to do a supervised transfer using this topology:pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk Because if I do a flash() and a SendDTMF() to transfer the extension I have to Hungup(), otherwise it never reaches the called extension. So if I do a Hangup() I cannot know of the result of the call.I think that the only way is going to be like this: pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk (fxo zap/2) - ATA - (ext 321) norstarWith that topology I'll be able to do a DIAL() on the other zap (zap/2) and with that know the state of the call. The problem with this topology is that for 5 lines is gonna be expensive and difficult to find 10 ATAs!If you have any suggestions and configurations they will be very appreciated.Thanks!-- Gustavo BermanSysadminDepto. InformaticaUniversidad Nacional del ComahueCentro Regional Universitario Bariloche ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and norstar
Hello Jorge, and thanks for the answers, but:I don't understand what is a blind transfer and a supervised transfer.I mean, in the topology:- pstn line - norstar (ext 123) - ATA - (fxo zap/1) asterisk
An incoming call from the pstn line is forwarded by the norstar to extension 123 were asterisk is.So asterisk answers the call and play a background message for the caller. But when the user enter the extension number what do we have to do?
I tried with:Hook flash version:exten = _XXX,1,Flash() ;do a hook flash (like pressing FUNCTION in meridian phone)exten = _XXX,2,SendDTMF(*70w${EXTEN},250) ;sends the code for transfer plus the extension
exten = _XXX,3,Hangup()In this version I can transfer the call using the same channel (zap/1) but didn't find a way for voicemail if the call is unanswered or is busy.Also if its unanswered the call is returned to the extension were asterisk is.
Dial version:exten = _XXX,1,Dial(ZAP/1/${EXTEN})It says the channel is busy.I think that with this version I can have a dialstatus for sending to voicemailSo, a couple of questions:What is a blind and a supervised transfer? (cannot find it in the norstar manual)
Do you have and use this topology? if so, how do you do it?Thanks for the help!!(I'm a linux sysadmin and never before worked with telephones system)-- Gustavo BermanSysadminDepto. Informatica
Universidad Nacional del ComahueCentro Regional Universitario Bariloche
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Re: [asterisk-users] announcing inbound PSTN calls
Interesting!I think this can help for a start (but I don't know how to continue!!):[incoming]exten = s,1,Answer()exten = s,2,Backgroud(enter-ext)exten = _XXX,1,Playback(enter-name)exten = _XXX,2,Record(/tmp/prompt${EXTEN}:wav)
exten = _XXX,3,Dial(zap/1/${EXTEN})now, how to play the recorded message to the called party when he/she answers the phone?any help?On 11/10/06,
Jeronimo Romero [EMAIL PROTECTED] wrote:
I'm running asterisk 1.2.8. I would like PSTN inbound
calls to do the following:
1-once PSTN callers enter their desired extension; they have
to record their name
2-recording then announces that it is trying to locate the
user
3-asterisk calls local extension and announces callers
recorded name
4-local recipient user can choose to take the call, send it
to voicemail or transfer it to another extension
Is this possible in asterisk?? . If it is possible, what is
the name of this function? Is this documented anywhere?
What is the best approach to doing this?
Thanks in advance
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-- Gustavo BermanSysadminDepto. InformaticaUniversidad Nacional del ComahueCentro Regional Universitario Bariloche
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[asterisk-users] asterisk and norstar
Hi there!We have an old legacy norstar phone system m8x24-ds ( dr5 ) and a couple of m0x16. It has 5 external analog lines. It has no auto attendant, and no voicemail. So every incoming call is forwarded to a operator, she pick up the phone, talks to the caller and transfer the call to the right extension. We are in Argentina, so buying a star talk is out of the question, there is no selling of that in here.So, we want to use * as an auto attendant and voicemail for our 50 extensions.Is there anybody who has done that? What topology do we have to use? :1) pstn line - (fxo) asterisk (fxs) - norstaror2) pstn line - (fxo/1) asterisk (fxo/2) - ata - (ext. 123) norstaror3) pstn line - norstar (ext. 123) - ata/1 - (fxo/1) asterisk (fxo/2) - ata/2 - ( ext.321) norstaror4) pstnl line - norstar (ext. 123) - ata - (fxo) asteriskAny help please?I'm not a telephone systems specialist!Thanks!-- Gustavo BermanSysadminDepto. Informatica Universidad Nacional del ComahueCentro Regional Universitario Bariloche ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
