Re: [Asterisk-Users] Why use 'Answer'?
Rich Adamson schrieb: Rephrased: Why do folks think they have to use Answer in the sequence when Playback (etc) is _not_ used? Because they don't think or they love the telephone companies ... I think its stupid because the call is established and the caller has to pay for it even when the call isn't really answered by a human or an answering system. Just imagine you make a call from a place with very high rates (satellite phone) and you let the phone ring several times maybe 60 sec before disconnecting because the called person isn't at place. This can be a very cost intensive experience ... Bye! Michael ___ 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] Why use 'Answer'?
On Thu, Dec 23, 2004 at 11:45:04AM +0100, Michael Vogel wrote: Rich Adamson schrieb: Rephrased: Why do folks think they have to use Answer in the sequence when Playback (etc) is _not_ used? Because they don't think or they love the telephone companies ... Ok, this is probably a stupid question ;) If I have a setup like the following: One TDM400P with one FXO interface in slot 4. zapata.conf chunk: signalling=fxs_ks language=en context=in5100 channel = 4 extensions.conf chunk: [in5100] exten = s,1,Wait,2 exten = s,n,Answer exten = s,n,DigitTimeout,5 exten = s,n,ResponseTimeout,10 exten = s,n(restart),BackGround(demo-congrats) exten = s,n,WaitExten exten = 401,1,Dial(SIP/sip1,20,tr) /// I'm just playing demo-congrats so I can hear something when I call in to know it's working. What I would really like to have happen for now is to ring the sip1 phone when the incoming line rings and only answer the incoming line if the sip1 phone gets answered. I've been playing with * for a while, but this is my first zap device :) [been all sip and iax til now] Clues? Regards, -Dorn ___ 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] Why use 'Answer'?
Rephrased: Why do folks think they have to use Answer in the sequence when Playback (etc) is _not_ used? Because they don't think or they love the telephone companies ... Ok, this is probably a stupid question ;) If I have a setup like the following: One TDM400P with one FXO interface in slot 4. zapata.conf chunk: signalling=fxs_ks language=en context=in5100 channel = 4 extensions.conf chunk: [in5100] exten = s,1,Wait,2 exten = s,n,Answer exten = s,n,DigitTimeout,5 exten = s,n,ResponseTimeout,10 exten = s,n(restart),BackGround(demo-congrats) exten = s,n,WaitExten exten = 401,1,Dial(SIP/sip1,20,tr) /// I'm just playing demo-congrats so I can hear something when I call in to know it's working. What I would really like to have happen for now is to ring the sip1 phone when the incoming line rings and only answer the incoming line if the sip1 phone gets answered. I've been playing with * for a while, but this is my first zap device :) [been all sip and iax til now] Clues? As in many cases with *, there are usually multiple ways to accomplish a task. Here's a couple that you'll need to tailor to your environment. [in5100] exten = s,1,Dial(SIP/sip1,20,tr) The above assumes the pstn line is _not_ sending any digits to you. If it does, then replace s with whatever they are sending to you. Or, if you don't want to maintain multiple instances of the Dial command, then do something like: [in5100] exten = s,1,Goto(local-extns,sip1,1) The above assumes your sip phone dialplan reside in the [local-extns] context, that sip1 is a valid extn number within that context, and 1 is the first priority in that sip phone definition. When someone answers the sip phone, the zap pstn line will be answered. If no one answers the sip phone, the 20 second timer will expire and ordinarily it would jump to the next priority. So, you probably want to remove the timeout altogether from that exten definition and just let it ring forever. ___ 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] Why use 'Answer'?
Thanks! :) On Thu, Dec 23, 2004 at 08:57:22AM -0600, Rich Adamson wrote: As in many cases with *, there are usually multiple ways to accomplish a task. Here's a couple that you'll need to tailor to your environment. [in5100] exten = s,1,Dial(SIP/sip1,20,tr) The above assumes the pstn line is _not_ sending any digits to you. If it does, then replace s with whatever they are sending to you. If the PSTN line is just an analog line connected to the TDM400P FXO interface, is there any way it will be sending me digits? Or, if you don't want to maintain multiple instances of the Dial command, then do something like: [in5100] exten = s,1,Goto(local-extns,sip1,1) The above assumes your sip phone dialplan reside in the [local-extns] context, that sip1 is a valid extn number within that context, and 1 is the first priority in that sip phone definition. so I would need something like: [local-extns] exten = sip1,1,Dial(SIP/sip1,,tr) ?? When someone answers the sip phone, the zap pstn line will be answered. If no one answers the sip phone, the 20 second timer will expire and ordinarily it would jump to the next priority. So, you probably want to remove the timeout altogether from that exten definition and just let it ring forever. If I want to timeout to voicemail then what would the line(s) in [local-extns] look like? -Dorn ___ 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] Why use 'Answer'?
Inline... As in many cases with *, there are usually multiple ways to accomplish a task. Here's a couple that you'll need to tailor to your environment. [in5100] exten = s,1,Dial(SIP/sip1,20,tr) The above assumes the pstn line is _not_ sending any digits to you. If it does, then replace s with whatever they are sending to you. If the PSTN line is just an analog line connected to the TDM400P FXO interface, is there any way it will be sending me digits? Not normally, however in past years telco's would provide DID over about any type of loop. If you didn't order DID or anything special, then you are not going to receive any digits from the telco. Or, if you don't want to maintain multiple instances of the Dial command, then do something like: [in5100] exten = s,1,Goto(local-extns,sip1,1) The above assumes your sip phone dialplan reside in the [local-extns] context, that sip1 is a valid extn number within that context, and 1 is the first priority in that sip phone definition. so I would need something like: [local-extns] exten = sip1,1,Dial(SIP/sip1,,tr) ?? When someone answers the sip phone, the zap pstn line will be answered. If no one answers the sip phone, the 20 second timer will expire and ordinarily it would jump to the next priority. So, you probably want to remove the timeout altogether from that exten definition and just let it ring forever. If I want to timeout to voicemail then what would the line(s) in [local-extns] look like? Something like: [in5100] exten = sip1,1,Dial(SIP/sip1,20,tr) exten = sip1,2,Voicemail(usip1) exten = sip1,102,Voicemail(bsip1) exten = sip1,103,Hangup ___ 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] Why use 'Answer'?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, December 22, 2004 8:12 AM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
I fully understand that incoming pstn calls have to be answered (in
most cases) before executing a Playback(invalid) type statement. But
there must be some examples, documentation, or somthing that is
suggesting to newcomers that all sequences have to start with an Answer.
Question:
Do you need to answer to detect a fax?
Excerpt from my conf file.
I would like to tune this. I have tried putting the 800 service checks on a
single line but it fails. Any advice would be useful.
[inbound-pstn]
exten = s,1,NoOp(${CALLERID}) ; log callerID string
exten = s,2,GotoIf($[${CALLERIDNUM:0:3} = 800]?s|108)
exten = s,3,GotoIf($[${CALLERIDNUM:0:3} = 866]?s|108)
exten = s,4,GotoIf($[${CALLERIDNUM:0:3} = 877]?s|108)
exten = s,5,GotoIf($[${CALLERIDNUM:0:3} = 888]?s|108)
exten = s,6,GotoIf($[${CALLERIDNUM} = ]?s|109)
exten = s,7,LookupBlacklist
exten = s,8,Answer
exten = s,9,Ringing
exten = fax,1,Macro(faxreceive)
exten = s,10,Macro(ringphones)
exten = s,11,Wait(2)
exten = s,12,PlayBack(im-sorry)
exten = s,13,Voicemail(u100)
exten = s,14,Hangup
exten = s,108,Macro(noservice)
exten = s,109,Macro(nocallerid)
exten = h,1,Hangup
--John
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RE: [Asterisk-Users] Why use 'Answer'?
On Wed, 2004-12-22 at 08:41, John Hill wrote: Question: Do you need to answer to detect a fax? Yes. You need to answer the line so the calling fax will start sending the fax tones and * can detect them. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Why use 'Answer'?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Wednesday, December 22, 2004 10:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Why use 'Answer'? On Wed, 2004-12-22 at 08:41, John Hill wrote: Question: Do you need to answer to detect a fax? Yes. You need to answer the line so the calling fax will start sending the fax tones and * can detect them. -Seth -- That's what I thought. My dial plan works but it looks a bit messy. --John ___ 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] Why use 'Answer'?
There's a wiki tip that suggests you always put answer and even wait
before playback, cause asterisk can pickup a milisec after you've
finished dialing, unlike legacy PBXs that always ring at least once.
Take a look at
http://voip-info.org/wiki-Asterisk+tips+answer-before-playback
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 22, 2004 3:12 PM
To: Asterisk-a-users-list
Subject: [Asterisk-Users] Why use 'Answer'?
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
I fully understand that incoming pstn calls have to be answered (in
most cases) before executing a Playback(invalid) type statement. But
there must be some examples, documentation, or somthing that is
suggesting to newcomers that all sequences have to start with an
Answer.
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RE: [Asterisk-Users] Why use 'Answer'?
That's an answer to the wrong question. See example below.
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
There's a wiki tip that suggests you always put answer and even wait
before playback, cause asterisk can pickup a milisec after you've
finished dialing, unlike legacy PBXs that always ring at least once.
Take a look at
http://voip-info.org/wiki-Asterisk+tips+answer-before-playback
-Original Message-
Why is it that newcomers always feel like inserting 'Answer' is a
necessary step in their extension.conf entries?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
I fully understand that incoming pstn calls have to be answered (in
most cases) before executing a Playback(invalid) type statement. But
there must be some examples, documentation, or somthing that is
suggesting to newcomers that all sequences have to start with an
Answer.
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Re: [Asterisk-Users] Why use 'Answer'?
Rich Adamson wrote:
That's an answer to the wrong question. See example below.
Rephrased: Why do folks think they have to use Answer in the sequence
when Playback (etc) is _not_ used?
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
Urban myth. There is no reason to use answer for the example above.
People just repeat what they have been told or have heard (I've done
that myself too occasionally) and so incorrect information gets passed
around.
In fact using Answer could very well prevent the caller from hearing
ringing. I don't know, I would have to look at a SIP debug.
In the example above the call will hangup if the destination is
unavailable OR busy. You usually want to handle these events by looking
at the DIALSTATUS variable. See the stdexten macro in
extensions.conf.sample.
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RE: [Asterisk-Users] Why use 'Answer'?
On Wed, 22 Dec 2004, Rich Adamson wrote: That's an answer to the wrong question. See example below. Rephrased: Why do folks think they have to use Answer in the sequence when Playback (etc) is _not_ used? And even if you do play back some audio you may not want to answer anyway. In most modern phone systems the reverse path is set up early so you can play a personalized busy message, a personalized ringing signal etc. It's a neat trick. Note that playback will answer the line unless given a special option. On the other hand, if you want to get *any* audio from the caller you need to answer the line. Some applications (such as dial) will answer an unanswered line before going on. Peter ___ 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
