asterisk users - Dec 2004 - How to create a confrence using SIP channels

Hello,

I would like to be able to dial in to my asterisk box.
Dial extension which would call two other people using the Sip channels.
We would like to be able to talk to each other at the same time.

Thanks

Bartosz Wegrzyn

On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
> I would like to be able to dial in to my asterisk box.
> Dial extension which would call two other people using the Sip channels.
> We would like to be able to talk to each other at the same time.
This is quite easy. :-)

Have the extension construct two call files and place them in the outgoing
call spool directory then proceed to the meetme room. The call files
should contain the meetme extension as well. Thus the caller will go to
the meetme conference directly and the two called parties will enter the 
conference as soon as they answer.

Peter

Can you show me the simple example of this in asterisk words?

Thanks
> On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>
>> I would like to be able to dial in to my asterisk box.
>> Dial extension which would call two other people using the Sip
channels.
>> We would like to be able to talk to each other at the same time.
>
> This is quite easy. :-)
>
> Have the extension construct two call files and place them in the outgoing
> call spool directory then proceed to the meetme room. The call files
> should contain the meetme extension as well. Thus the caller will go to
> the meetme conference directly and the two called parties will enter the
> conference as soon as they answer.
>
> Peter
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
> Can you show me the simple example of this in asterisk words?
Make an extension that calls an agi and then starta the MeetMe application 
in Asterisk. The agi should create two call files (see the wiki for 
details such as moving the *complete* file into the spool directory to 
avoid races) looking like this:

-Begin call file -----------------
Channel: Sip/?????????
Callerid: ????????      (sent as callerid to the channel)
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: auto_meetme
Extension: s
Priority: 1
-End call file   -----------------

Then have a context 
[auto_meetme]
exten => s,1,Meetme(1234)

You may want to use dynamic meetme rooms etc. This can be refined a lot. I 
have not tested the above, there are almost certainly bugs. It should get 
you started though.

Peter
> > On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
> >
> >> I would like to be able to dial in to my asterisk box.
> >> Dial extension which would call two other people using the Sip
channels.
> >> We would like to be able to talk to each other at the same time.
> >
> > This is quite easy. :-)
> >
> > Have the extension construct two call files and place them in the
outgoing
> > call spool directory then proceed to the meetme room. The call files
> > should contain the meetme extension as well. Thus the caller will go
to
> > the meetme conference directly and the two called parties will enter
the
> > conference as soon as they answer.
> >
> > Peter

Thanks a lot Peter, I will check this.
> On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>
>> Can you show me the simple example of this in asterisk words?
>
> Make an extension that calls an agi and then starta the MeetMe application
> in Asterisk. The agi should create two call files (see the wiki for
> details such as moving the *complete* file into the spool directory to
> avoid races) looking like this:
>
> -Begin call file -----------------
> Channel: Sip/?????????
> Callerid: ????????      (sent as callerid to the channel)
> MaxRetries: 1
> RetryTime: 60
> WaitTime: 30
> Context: auto_meetme
> Extension: s
> Priority: 1
> -End call file   -----------------
>
> Then have a context
> [auto_meetme]
> exten => s,1,Meetme(1234)
>
> You may want to use dynamic meetme rooms etc. This can be refined a lot. I
> have not tested the above, there are almost certainly bugs. It should get
> you started though.
>
> Peter
>
>> > On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>> >
>> >> I would like to be able to dial in to my asterisk box.
>> >> Dial extension which would call two other people using the Sip
>> channels.
>> >> We would like to be able to talk to each other at the same
time.
>> >
>> > This is quite easy. :-)
>> >
>> > Have the extension construct two call files and place them in the
>> outgoing
>> > call spool directory then proceed to the meetme room. The call
files
>> > should contain the meetme extension as well. Thus the caller will
go
>> to
>> > the meetme conference directly and the two called parties will
enter
>> the
>> > conference as soon as they answer.
>> >
>> > 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
>

I have read more about AGI and I thing that I get the idea.
I did not realize that the programming would be involved.
Is this the only way to do it?

Thanks
> Thanks a lot Peter, I will check this.
>
>> On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>
>>> Can you show me the simple example of this in asterisk words?
>>
>> Make an extension that calls an agi and then starta the MeetMe
>> application
>> in Asterisk. The agi should create two call files (see the wiki for
>> details such as moving the *complete* file into the spool directory to
>> avoid races) looking like this:
>>
>> -Begin call file -----------------
>> Channel: Sip/?????????
>> Callerid: ????????      (sent as callerid to the channel)
>> MaxRetries: 1
>> RetryTime: 60
>> WaitTime: 30
>> Context: auto_meetme
>> Extension: s
>> Priority: 1
>> -End call file   -----------------
>>
>> Then have a context
>> [auto_meetme]
>> exten => s,1,Meetme(1234)
>>
>> You may want to use dynamic meetme rooms etc. This can be refined a
lot.
>> I
>> have not tested the above, there are almost certainly bugs. It should
>> get
>> you started though.
>>
>> Peter
>>
>>> > On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>> >
>>> >> I would like to be able to dial in to my asterisk box.
>>> >> Dial extension which would call two other people using the
Sip
>>> channels.
>>> >> We would like to be able to talk to each other at the same
time.
>>> >
>>> > This is quite easy. :-)
>>> >
>>> > Have the extension construct two call files and place them in
the
>>> outgoing
>>> > call spool directory then proceed to the meetme room. The call
files
>>> > should contain the meetme extension as well. Thus the caller
will go
>>> to
>>> > the meetme conference directly and the two called parties will
enter
>>> the
>>> > conference as soon as they answer.
>>> >
>>> > 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
>>
>
> _______________________________________________
> 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
>

I have another simpler idea.
I create an extension that uses the system command to move files to
outgoing directory. After then it calls the meetme extension.

This would not require any programming, but this would also work for
specific numers. (hardcoded in the script)
But I can also pass arguments to that script and specifiy the number I
will call. For people like me (not programmers) this would be much easier.

Bart,
> I have read more about AGI and I thing that I get the idea.
> I did not realize that the programming would be involved.
> Is this the only way to do it?
>
> Thanks
>
>> Thanks a lot Peter, I will check this.
>>
>>> On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>>
>>>> Can you show me the simple example of this in asterisk words?
>>>
>>> Make an extension that calls an agi and then starta the MeetMe
>>> application
>>> in Asterisk. The agi should create two call files (see the wiki for
>>> details such as moving the *complete* file into the spool directory
to
>>> avoid races) looking like this:
>>>
>>> -Begin call file -----------------
>>> Channel: Sip/?????????
>>> Callerid: ????????      (sent as callerid to the channel)
>>> MaxRetries: 1
>>> RetryTime: 60
>>> WaitTime: 30
>>> Context: auto_meetme
>>> Extension: s
>>> Priority: 1
>>> -End call file   -----------------
>>>
>>> Then have a context
>>> [auto_meetme]
>>> exten => s,1,Meetme(1234)
>>>
>>> You may want to use dynamic meetme rooms etc. This can be refined a
>>> lot.
>>> I
>>> have not tested the above, there are almost certainly bugs. It
should
>>> get
>>> you started though.
>>>
>>> Peter
>>>
>>>> > On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>>> >
>>>> >> I would like to be able to dial in to my asterisk box.
>>>> >> Dial extension which would call two other people using
the Sip
>>>> channels.
>>>> >> We would like to be able to talk to each other at the
same time.
>>>> >
>>>> > This is quite easy. :-)
>>>> >
>>>> > Have the extension construct two call files and place them
in the
>>>> outgoing
>>>> > call spool directory then proceed to the meetme room. The
call files
>>>> > should contain the meetme extension as well. Thus the
caller will go
>>>> to
>>>> > the meetme conference directly and the two called parties
will enter
>>>> the
>>>> > conference as soon as they answer.
>>>> >
>>>> > 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
>>>
>>
>> _______________________________________________
>> 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
>

How should my outgoning spool file look like in order to call
using the sip channel (in this example using the Nikotel account)

I tried this, but this is not working.
Channel: Sip/1234566@Nikotel
Callerid: 1
MaxRetries: 1
RetryTime: 60
WaitTime: 30
Context: common
Extension: 500
Priority: 1

500 is my meetme extension.
Later,I want to be able to put more files to the outgoing folder.

Thanks

> I have another simpler idea.
> I create an extension that uses the system command to move files to
> outgoing directory. After then it calls the meetme extension.
>
> This would not require any programming, but this would also work for
> specific numers. (hardcoded in the script)
> But I can also pass arguments to that script and specifiy the number I
> will call. For people like me (not programmers) this would be much easier.
>
> Bart,
>
>> I have read more about AGI and I thing that I get the idea.
>> I did not realize that the programming would be involved.
>> Is this the only way to do it?
>>
>> Thanks
>>
>>> Thanks a lot Peter, I will check this.
>>>
>>>> On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>>>
>>>>> Can you show me the simple example of this in asterisk
words?
>>>>
>>>> Make an extension that calls an agi and then starta the MeetMe
>>>> application
>>>> in Asterisk. The agi should create two call files (see the wiki
for
>>>> details such as moving the *complete* file into the spool
directory to
>>>> avoid races) looking like this:
>>>>
>>>> -Begin call file -----------------
>>>> Channel: Sip/?????????
>>>> Callerid: ????????      (sent as callerid to the channel)
>>>> MaxRetries: 1
>>>> RetryTime: 60
>>>> WaitTime: 30
>>>> Context: auto_meetme
>>>> Extension: s
>>>> Priority: 1
>>>> -End call file   -----------------
>>>>
>>>> Then have a context
>>>> [auto_meetme]
>>>> exten => s,1,Meetme(1234)
>>>>
>>>> You may want to use dynamic meetme rooms etc. This can be
refined a
>>>> lot.
>>>> I
>>>> have not tested the above, there are almost certainly bugs. It
should
>>>> get
>>>> you started though.
>>>>
>>>> Peter
>>>>
>>>>> > On Mon, 13 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
>>>>> >
>>>>> >> I would like to be able to dial in to my asterisk
box.
>>>>> >> Dial extension which would call two other people
using the Sip
>>>>> channels.
>>>>> >> We would like to be able to talk to each other at
the same time.
>>>>> >
>>>>> > This is quite easy. :-)
>>>>> >
>>>>> > Have the extension construct two call files and place
them in the
>>>>> outgoing
>>>>> > call spool directory then proceed to the meetme room.
The call
>>>>> files
>>>>> > should contain the meetme extension as well. Thus the
caller will
>>>>> go
>>>>> to
>>>>> > the meetme conference directly and the two called
parties will
>>>>> enter
>>>>> the
>>>>> > conference as soon as they answer.
>>>>> >
>>>>> > 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
>>>>
>>>
>>> _______________________________________________
>>> 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
>>
>
> _______________________________________________
> 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
>