Re: [Asterisk-Users] 2nd Dialtone after answer
Hi all, DID is Direct Inward Dial where the last 3 or 4 digits of the dialed number are passed through and are used/translated to call a specific extension. (See Centrex) DISA is Direct System Access where incoming line(s)are auto-answered and receive internal dial tone, the caller then has access to the facilities of the system.(including calling an extension.) I hope this clears things up TTFN Henry Chris Coulthurst wrote: Check out DISA. Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Oswaldo Arratia |Sent: Friday, June 17, 2005 7:51 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] 2nd Dialtone after answer | | |Hi |I am trying to achive this for a specific need of a customer. | |He has a DID pointed to an Asterisk server, I need to provide |him dialtone when the calls hits the server. How can I achieve this? | |Let's say something like this: | |Exten = s,1,Answer |Exten = s,2, Provide Dial tone |Exten = s,3, Dial the number the person will enter after |receiving the dial tone Exten = s,4,Hangup | |Any ideas? | |Thanks very much | |Oswaldo | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asteri|sk-users |To |UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adtran TA 750 FXO Groundstart Mode
I don't have experience of the FXO card but to start a ground start trunk you must have a good ground ( 2ohms) Connecting up a grounding strap around a copper cold water pipe works well. Henry Syed Akbar wrote: I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle one incoming call on multiple lines?
This feature is called attendant - night answer position. Is it not possible to switch the incoming call to an alternate extension based on time of day ? Henry Florian Overkamp wrote: Hi, -Original Message- Thanks, but it isn't an option because the Telco is actually connected to a PBX which is connected to Asterisk which should act as a intelligent answering device during non-office hours. The PBX isn't capable of doing this. Any other option? Hmm, this is a bit of a hack, but it might suit your needs: - Make sure each of those lines goes into a different extension or context - Add a delay on each line, like this: exten = line1,1,Do stuff exten = line2,1,Wait(2) exten = line2,1,Do stuff exten = line3,1,Wait(4) exten = line3,1,Do stuff exten = line4,1,Wait(6) exten = line4,1,Do stuff Could this help your case ? Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle one incoming call on multiple lines?
Yeh, this is called line hunting all telco's offer this... you get one published number but say 12 lines each line actually has a number but just calling the main number will automatically roll-over to the first available line in that hunting group. By the way, outgoing calls that use the same lines should have hunting groups going in the opposite direction (for obvious reasons). Unfortunatly, for those who want to develop ACD (Automatic Call Distribution) this mode is useless, if you were to distribute calls based on this method the person attached to the first line would get most of the calls while the last would be able to put their feet up and whistle dixie Have fun ..Henry Erwin Lubbers wrote: Julian, Thanks, but it isn't an option because the Telco is actually connected to a PBX which is connected to Asterisk which should act as a intelligent answering device during non-office hours. The PBX isn't capable of doing this. Any other option? Regards, Erwin Isn't it easier to talk to your Telco, and tell them to just ring the first free line, instead of all 4? Julian J. M. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to handle one incoming call on multiple lines?
Will do ..Thanks Henry Andrew Kohlsmith wrote: On Wednesday 08 June 2005 11:24, Henry Coleman wrote: This feature is called attendant - night answer position. Is it not possible to switch the incoming call to an alternate extension based on time of day ? You need to read up. This exact situation is given in the Asterisk Handbook. http://www.digium.com/handbook-draft.pdf In particular, you want GotoIfTime(). -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK
Hi Angus, If you connect the phone directly to the outside line will it ring ? The ring from the C.O. provides a 90volt AC (30cps) and is capable of ringing a standard phone ( a real two tone gong bell) My guess is that the TDM400 card does not supply enough current to actually do this. Most modern phones have an electronic ringer which requires a fraction of the power and will work fine. I don't quite understand the reference to a capacitor unless your phone is as old as I am in which case the phone has 2 x pairs of wires going to the phone plug. The first pair of wires are the voice pair and the second pair are connected to the ringer if this is the case your phone will work normally but simply doesn't ring. The fix is to connect two capacitors approx *0.15 uf 250vw *from each wire of the voice pair to each wire of the ring pair (you can do this inside the phone jack) This should not cost much (about a dollar) and can be found in any electronics component shop (try Maplin electronics) . Concidering the time and effort you might want to buy a new phone. 0.1uf Wall Jack TDM400 | ---||---yellow---0 0---green--0---0 0---red-0--- 0 | ---||---black-0 0.1uf Hope this helps ...Henry * * Angus Comber wrote: Hello I have played about with a TDM400 card and plugged in some standard analog phones. I am using the card in FXS mode - for analog extensions. I did notice that one of my phones did not ring and I wondered why. I later read in Paul Mahler's book VoIP Telephony with Asterisk that in his section on the TDM400 on page 127 he says In the UK, you may need an adapter that provides a ring capacitor, or the phone may not ring. Can anyone confirm this. Also what is one of those and where would I find a good supplier? I am in the trade so wholesale would be OK. Angus Comber ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
