[asterisk-users] Test

[Matt Fredrickson]

Testing, 1, 2, 3.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] invite to conference by a call file

[Frank Vanoni]

Here I'm using the "Page" application to make a conference call "on the
fly".



[office]

exten => ,1,Dial(SIP/desk2,150)
   same => n,Hangup()

exten => ,1,Dial(SIP/desk3,150)
   same => n,Hangup()

exten => ,1,Dial(SIP/desk4,150)
   same => n,Hangup()

exten => ,1,Dial(SIP/desk5,150)
   same => n,Hangup()

exten => ,1,Dial(SIP/desk6,150)
   same => n,Hangup()

; Conference call
exten => ,1,Answer
exten => ,n,Page(Local/@office/@office/@off
ice/@office/@office,d)
same => n,Hangup()-- 
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Re: [asterisk-users] invite to conference by a call file

[Dovid Bender]

You would make the call file the same way you are now. 100 was the conf
room ID. Have a look at the documentation how to do it. Also take a look at
the default settings in confbridge.conf

voice1*CLI> core show application ConfBridge

  -= Info about application 'ConfBridge' =-

[Synopsis]
Conference bridge application.

[Description]
Enters the user into a specified conference bridge.  The user can exit the
conference by hangup or DTMF menu option.
This application sets the following channel variable upon completion:
${CONFBRIDGE_RESULT}:
FAILED:The channel encountered an error and could not enter the
conference.
HANGUP:The channel exited the conference by hanging up.
KICKED:The channel was kicked from the conference.
ENDMARKED:The channel left the conference as a result of the last marked
user leaving.
DTMF:The channel pressed a DTMF sequence to exit the conference.
TIMEOUT:The channel reached its configured timeout.

[Syntax]
ConfBridge(conference[,bridge_profile[,user_profile[,menu]]])

[Arguments]
conference
Name of the conference bridge.  You are not limited to just numbers.
bridge_profile
The bridge profile name from confbridge.conf.  When left blank, a
dynamically built bridge profile created by the CONFBRIDGE dialplan
function is searched for on the channel and used.  If no dynamic
profile is
present, the 'default_bridge' profile found in confbridge.conf is used.
It is important to note that while user profiles may be unique for each
participant, mixing bridge profiles on a single conference is _NOT_
recommended and will produce undefined results.
user_profile
The user profile name from confbridge.conf.  When left blank, a
dynamically
built user profile created by the CONFBRIDGE dialplan function is
searched
for on the channel and used.  If no dynamic profile is present, the
'default_user' profile found in confbridge.conf is used.
menu
The name of the DTMF menu in confbridge.conf to be applied to this
channel.
 When left blank, a dynamically built menu profile created by the
CONFBRIDGE dialplan function is searched for on the channel and used.
If no
dynamic profile is present, the 'default_menu' profile found in
confbridge.conf is used.

[See Also]
ConfBridge(), CONFBRIDGE, CONFBRIDGE_INFO

On Tue, Mar 20, 2018 at 10:44 AM, Atux Atux  wrote:

> thanks a lot for the reply.
> [call-file-test]
> Exten => 10,1,Answer
>  same => ConfBridge(100)
>
>
> i assume 100 is the conference room, correct?
> where do i write the SIP numbers to invite(internal or external)?
> what about the PIN?
>
>
> On Tue, Mar 20, 2018 at 4:41 PM, Dovid Bender  wrote:
>
>> Atux,
>>
>> This should work:
>> [call-file-test]
>> Exten => 10,1,Answer
>>  same => ConfBridge(100)
>>
>> On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux  wrote:
>>
>>> Hi. in my system i have a conference room where someone can call it eg
>>> 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in
>>> through a different number and PIN.  I would like to have a call file and
>>> call all participants eg 610-619 at certain time of the day and give them
>>> access to the conference.
>>> During my try i managed to create a call file where it calls the a SIP
>>> phone and it can hear the monkeys (just for test).
>>> here is the call file
>>> Channel: SIP/601
>>> MaxRetries: 2
>>> RetryTime: 60
>>> WaitTime: 30
>>> Context: call-file-test
>>> Extension: 10
>>>
>>>
>>>
>>> and here is the entry in extensions.conf
>>>
>>> [call-file-test]
>>> exten => 10,1,Answer()
>>> exten => 10,n,Wait(1)
>>> exten => 10,n,Playback(tt-monkeys)
>>> exten => 10,n,Wait(1)
>>> exten => 10,n,Hangup()
>>>
>>>
>>> i did not manage to make it call more SIP phones and invite them to the
>>> conference
>>>
>>> Any ideas please?
>>>
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>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
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>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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Re: [asterisk-users] invite to conference by a call file

[Atux Atux]

thanks a lot for the reply.
[call-file-test]
Exten => 10,1,Answer
 same => ConfBridge(100)


i assume 100 is the conference room, correct?
where do i write the SIP numbers to invite(internal or external)?
what about the PIN?


On Tue, Mar 20, 2018 at 4:41 PM, Dovid Bender  wrote:

> Atux,
>
> This should work:
> [call-file-test]
> Exten => 10,1,Answer
>  same => ConfBridge(100)
>
> On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux  wrote:
>
>> Hi. in my system i have a conference room where someone can call it eg
>> 698 dial the PIN eg 1234 and enter the room as a user. The admin enters in
>> through a different number and PIN.  I would like to have a call file and
>> call all participants eg 610-619 at certain time of the day and give them
>> access to the conference.
>> During my try i managed to create a call file where it calls the a SIP
>> phone and it can hear the monkeys (just for test).
>> here is the call file
>> Channel: SIP/601
>> MaxRetries: 2
>> RetryTime: 60
>> WaitTime: 30
>> Context: call-file-test
>> Extension: 10
>>
>>
>>
>> and here is the entry in extensions.conf
>>
>> [call-file-test]
>> exten => 10,1,Answer()
>> exten => 10,n,Wait(1)
>> exten => 10,n,Playback(tt-monkeys)
>> exten => 10,n,Wait(1)
>> exten => 10,n,Hangup()
>>
>>
>> i did not manage to make it call more SIP phones and invite them to the
>> conference
>>
>> Any ideas please?
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] invite to conference by a call file

[Dovid Bender]

Atux,

This should work:
[call-file-test]
Exten => 10,1,Answer
 same => ConfBridge(100)

On Tue, Mar 20, 2018 at 10:34 AM, Atux Atux  wrote:

> Hi. in my system i have a conference room where someone can call it eg 698
> dial the PIN eg 1234 and enter the room as a user. The admin enters in
> through a different number and PIN.  I would like to have a call file and
> call all participants eg 610-619 at certain time of the day and give them
> access to the conference.
> During my try i managed to create a call file where it calls the a SIP
> phone and it can hear the monkeys (just for test).
> here is the call file
> Channel: SIP/601
> MaxRetries: 2
> RetryTime: 60
> WaitTime: 30
> Context: call-file-test
> Extension: 10
>
>
>
> and here is the entry in extensions.conf
>
> [call-file-test]
> exten => 10,1,Answer()
> exten => 10,n,Wait(1)
> exten => 10,n,Playback(tt-monkeys)
> exten => 10,n,Wait(1)
> exten => 10,n,Hangup()
>
>
> i did not manage to make it call more SIP phones and invite them to the
> conference
>
> Any ideas please?
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] invite to conference by a call file

[Atux Atux]

Hi. in my system i have a conference room where someone can call it eg 698
dial the PIN eg 1234 and enter the room as a user. The admin enters in
through a different number and PIN.  I would like to have a call file and
call all participants eg 610-619 at certain time of the day and give them
access to the conference.
During my try i managed to create a call file where it calls the a SIP
phone and it can hear the monkeys (just for test).
here is the call file
Channel: SIP/601
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: call-file-test
Extension: 10



and here is the entry in extensions.conf

[call-file-test]
exten => 10,1,Answer()
exten => 10,n,Wait(1)
exten => 10,n,Playback(tt-monkeys)
exten => 10,n,Wait(1)
exten => 10,n,Hangup()


i did not manage to make it call more SIP phones and invite them to the
conference

Any ideas please?
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Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

[Benoit Panizzon]

Hi Tryba

> A (very) dirty workaround would be to drop these packets with iptables
> (assuming Linux as OS), something like:
> 
> iptables -t raw -I OUTPUT -p udp -d ipaddrofpbx -m string --algo bm
> --from 0 --to 32 --string "SIP/2.0 100 " -j DROP
> 
> Don't try it with TCP :)

:-)

Indeed, this is what I did with a Mikrotik Firewall that is in front
of the * Server: Drop UDP packet with content starting with "SIP/2.0
100 Trying"

And this showed, that not the missing 100 Trying is tripping our SBC.
So I contacted the Vendor who had a quick look at the 181 Ringing which
is not being relayed and found that the issue was the Contact: Header
containing two line= attributes.

So I'm now trying to contact the vendor of that other PBX to figure
out, why it is sending two line= attributes as part of the contact
header.

Mit freundlichen Grüssen

-Benoît Panizzon-
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Re: [asterisk-users] Is 100 trying mandatory? Can asterisk answer with 180 without prior 100 trying?

[Daniel Tryba]

On Mon, Mar 19, 2018 at 12:59:47PM -0300, Joshua Colp wrote:
> > To try to reproduce the problem with our SBC, is there a way to tell
> > the asterisk, preferably PJSIP, to directly answer with 180 ringing
> > without prior 100 trying?
> 
> The PJSIP channel driver has no option or ability to do this. I do not recall 
> if chan_sip does.

A (very) dirty workaround would be to drop these packets with iptables
(assuming Linux as OS), something like:

iptables -t raw -I OUTPUT -p udp -d ipaddrofpbx -m string --algo bm  --from 0 
--to 32 --string "SIP/2.0 100 " -j DROP

Don't try it with TCP :)


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