RE: [Asterisk-Users] Extensions to solve three way calling problem
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
The meetme option is nice, but it doesn't solve the problem. The TDM11B
only
has one FXO, one FXS. To get the effect the daughter wants requires
supporting the threeway facility the telco offers. You need to Flash the
outside line. Zap does have an application for that, but I haven't played
with what it can do, or how to program it.
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash() ;
flash the pstn line
What makes you think that would flash the PSTN line? This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection.
Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Extensions to solve three way calling problem
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold
message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash()
;
flash the pstn line
What makes you think that would flash the PSTN line?
Because the cli reports that it is executing flash on the Zap/4 - the PSTN
line
This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection.
Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
Yeah, I think you are right.
But what is the point of threewaycalling and transfer in zapata.conf - what
do they do?
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RE: [Asterisk-Users] Extensions to solve three way calling problem
On Wed, 2005-01-05 at 19:27 +1100, PHP Mechanic wrote:
On Wed, 2005-01-05 at 18:38 +1100, PHP Mechanic wrote:
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold
message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash()
;
flash the pstn line
What makes you think that would flash the PSTN line?
Because the cli reports that it is executing flash on the Zap/4 - the PSTN
line
This is your
problem. When you transfer the PSTN line anywhere and then go to dial
again, the flash is actually on the current channel. I wouldn't be
surprised if you hear it in your receiver. I don't know of anyway to
flash the PSTN line from within asterisk that would do as you want. In
fact, to enable it would be a security risk as well. Think of the
possibility of having multiple lines in and then dialing an extension to
flash the line and messing up and flashing someone else's connection.
Closest thing I could think of is having your PSTN side caller do the
transfer and redial. If the PSTN caller was allowed to transfer the
inside person and then dial a special extension that would initiate the
flash and the dial command. Of course the trouble here is that as soon
as the flash occurs, the new caller is the one going to be stuck in an
odd state and the previous PSTN caller is going to be in unrecoverable
limbo.
Just looks like you will be SOL on utilizing the PSTN 3 way calling.
Yeah, I think you are right.
But what is the point of threewaycalling and transfer in zapata.conf - what
do they do?
All of it is for doing stuff within asterisk. For example transfer is
for if you have more than one station inside the PBX, then you could
transfer the call from one phone to the other.
Threeway calling is similar. You can make a small impromptu conference
that way with 2 internal phones and an external or 3 internal phones or
even 1 internal and 2 external calls on separate phone lines. All of
these are mixed inside of asterisk and the PSTN is non the wiser.
--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Extensions to solve three way calling problem
Threeway calling is similar. You can make a small impromptu conference that way with 2 internal phones and an external or 3 internal phones or even 1 internal and 2 external calls on separate phone lines. All of these are mixed inside of asterisk and the PSTN is non the wiser. Thanks for clearing this up for me. The thing that still get's me is that the pstn is a little bit wise. It can perform the following: 1. Establish a call with the first person. You can call them or they can call you. 2. Press Flash/Recall on phone to put the first person on hold. 3. Wait until you hear the dial tone. 4. Dial the number of the second person. 5. Wait until you hear the second line ringing. 6. Press Flash/Recall and talk to the first person (they will hear the ringing tone too). Can I make asterisk play ball with my telco? ___ 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] Extensions to solve three way calling problem
On Wed, 5 Jan 2005 07:54:43 +0100, Florian Overkamp wrote: Hi, -Original Message- Have you considered setting up a meetme confrence line for them? :) analog phone = asterisk/tdm11b = pstn The meetme option is nice, but it doesn't solve the problem. The TDM11B only has one FXO, one FXS. To get the effect the daughter wants requires supporting the threeway facility the telco offers. You need to Flash the outside line. Zap does have an application for that, but I haven't played with what it can do, or how to program it. *CLI show application Flash -= Info about application 'Flash' =- [Synopsis]: Flashes a Zap Trunk [Description]: Flash(): Sends a flash on a zap trunk. This is only a hack for people who want to perform transfers and such via AGI and is generally quite useless otherwise. Returns 0 on success or -1 if this is not a zap trunk I've never bothered with Flash myself. I'd setup an account with an ITSP like VoipJet or Sixtel. Then you can initial the calls from in-house and make multiple outgoing connections, transfering each into meetme. Then again, if you used a sip phone (as opposed to an ata or TDM) you could conference one the phone withour resorting to meetme. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ 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] Extensions to solve three way calling problem
Have you considered setting up a meetme confrence line for them? :) On Wed, 5 Jan 2005 11:49:40 +1100, PHP Mechanic [EMAIL PROTECTED] wrote: My daughter hates my phone system because she can't use the three way function. What should my extension.conf look like to solve this problem. Without asterisk I can do the following without any trouble. 1. Establish a call with the first person. You can call them or they can call you. 2. Press Three Way, or Flash/Recall on any other touch tone phone to put the first person on hold. 3. Wait until you hear the dial tone. 4. Dial the number of the second person. 5. Wait until you hear the second line ringing. 6. Press Three Way or Flash/Recall and talk to the first person (they will hear the ringing tone too). However I have the following setup and I want to be able to do the same thing and it just doesn't work. analog phone = asterisk/tdm11b = pstn Can this be done? Does anyone have some extensions they could share with me which could achieve the same thing. ___ 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] Extensions to solve three way calling problem
From: Dr. Matthew Roller Have you considered setting up a meetme confrence line for them? :) I've tried that and I get as far as flashing the pstn after a transfer: ie: Hit the '#' and then '1' on the analogue phone. exten = 1,1,Flash() exten =1,2,MeetMe,2000 The problem is that after transfering the first caller to MeetMe I am unable to make an outgoing call using the pstn line, I don't get the dialtone and if I hang up and dial out I get a everyone is busy message, because I'm assuming that the line is still tied up with the first caller who is now in MeetMe. This problem has been bugging me for weeks. It's possible to do without using asterisk and all the settings in zapata.conf seem to suggest it's possible. But I am unable to make it work. Any other suggestions welcome. ___ 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] Extensions to solve three way calling problem
Hi, -Original Message- Have you considered setting up a meetme confrence line for them? :) analog phone = asterisk/tdm11b = pstn The meetme option is nice, but it doesn't solve the problem. The TDM11B only has one FXO, one FXS. To get the effect the daughter wants requires supporting the threeway facility the telco offers. You need to Flash the outside line. Zap does have an application for that, but I haven't played with what it can do, or how to program it. *CLI show application Flash -= Info about application 'Flash' =- [Synopsis]: Flashes a Zap Trunk [Description]: Flash(): Sends a flash on a zap trunk. This is only a hack for people who want to perform transfers and such via AGI and is generally quite useless otherwise. Returns 0 on success or -1 if this is not a zap trunk Best regards, 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
RE: [Asterisk-Users] Extensions to solve three way calling problem
Have you considered setting up a meetme confrence line for them? :)
analog phone = asterisk/tdm11b = pstn
The meetme option is nice, but it doesn't solve the problem. The TDM11B
only
has one FXO, one FXS. To get the effect the daughter wants requires
supporting the threeway facility the telco offers. You need to Flash the
outside line. Zap does have an application for that, but I haven't played
with what it can do, or how to program it.
I have played with it. But the problem I'm having is as follows
exten = _1800.,1,Dial(Zap/4/${EXTEN},20,Tr) ; call some company
willing to pay for my test, preferably get someone with an on hold message
; Now I press #* on the analog phone to transfer them to Meetme
exten = *,1,Meetme,2000 ; send
them to meetme
exten = *,2,Flash() ;
flash the pstn line
At this point they are moved to meetme and the pstn line is flashed.
I am disconnected and don't get a dialtone, I get a busy tone.
So I physically hang up the phone and then attempt to call another 1800
number
When I do this the cli displays a notice:
Unable to create channel of type 'Zap' == Everyone is busy/congested at
this time
It is possible for me to join the meetme by dialing 2000 but I can't appear
to access the pstn line that I flashed
What I don't understand is that there are settings in zapata.conf for
threewaycalling, transfer, etc... but it doesn't seem to work and I wonder
what these settings do? But more importantly is there a way to solve my
problem?
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