Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 06.11.2021 um 21:15 schrieb Łukasz Grzywański:

Hi Łukasz,

Dziękuję

> two legs in this same context
> ( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) )
> 
> PJSIP/pbxmichael_in-0418
> and 
> Local/123456@main_incoming-0268
> 
> [main_incoming]
> exten => _+49X.,1,goto(${EXTEN:3},1)
> exten => _0049X.,1,goto(${EXTEN:4},1)
> exten => _03529X.,1,goto(${EXTEN:1},1)
> exten => _3529X.,1,goto(${EXTEN:4},1)
> 
> exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
> exten => _123456,n,Set(CHANNEL(musicclass)=default)
> exten => _123456,n,Dial(SIP/74,39,RcxX)
> exten => _123456,n,Verbose(2,Voicemail for Main)
> exten => _123456,n,Set(CALLERID(name)=)
> exten => _123456,n,VoiceMail(74,us)
> exten => _123456,n,Hangup

You are my hero!
It works as expected!

Thank you very very much!
Luca

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Re: [asterisk-users] Notifying missed calls

[Łukasz Grzywański]

Hi ;-)
now I see 

Luca
two legs in this same context
( exten => _03529123456,n,Dial(local/123456@main_incoming,,xX) )

PJSIP/pbxmichael_in-0418
and
Local/123456@main_incoming-0268

[main_incoming]
exten => _+49X.,1,goto(${EXTEN:3},1)
exten => _0049X.,1,goto(${EXTEN:4},1)
exten => _03529X.,1,goto(${EXTEN:1},1)
exten => _3529X.,1,goto(${EXTEN:4},1)

exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" = "+49"
]?0${CALLERID(num):3}:${CALLERID(num)})})
exten => _123456,n,Set(CHANNEL(musicclass)=default)
exten => _123456,n,Dial(SIP/74,39,RcxX)
exten => _123456,n,Verbose(2,Voicemail for Main)
exten => _123456,n,Set(CALLERID(name)=)
exten => _123456,n,VoiceMail(74,us)
exten => _123456,n,Hangup

LG
Lukasz


On Sat, 6 Nov 2021 at 21:02, Łukasz Grzywański 
wrote:

> Hi,
> strange
>
>   -- Goto (noanswer,s,1)
>   -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418",
> "echo "Verpasster Anruf vom +4935 um 19:13" | mail -s "Verpasster
> Anruf" i...@mydomain.de") in new stack
>   -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-0268;2",
> "UID CALL: 1636222382.6032/ DATE: 20211106-191306)") in new stack
>   -- Executing [s@noanswer:2] System("Local/123456@main_incoming-0268;2",
> "echo "Verpasster Anrufvom 035 um 19:13" | mail -s "Verpasster
> Anruf" i...@mydomain.de") in new stack
>
> pls run
>
> asterisk -rx "dialplan show noanswer"
>
> and please check:
>
> [noanswer]
> exten => s,1,NoOp(UID CALL: ${UNIQUEID} /
> DATE:${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
> exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
> ${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@.de)
> exten => s,n,Hangup()
>
>
>  LG
> Lukasz
>
> On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello 
> wrote:
>
>> Am 06.11.2021 um 15:06 schrieb Frank Vanoni:
>>
>> Hi Frank
>>
>> > The "h" extension is executed whenever a call is hang up in that
>> > contexts.
>> >
>> > In your configuration it executes first the "s" extension (where you
>> > GoTo h,1) and once that is executed, the "h" extension is executed
>> > again.
>>
>> OK, I modified my configuration so:
>>
>> [main_incoming]
>> exten => _00493529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
>> exten => _00493529123456,n,Dial(local/123456@main_incoming,,xX)
>> exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
>> exten => _03529123456,n,Dial(local/123456@main_incoming,,xX)
>> exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
>> exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
>> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
>> exten => _123456,n,Set(CHANNEL(musicclass)=default)
>> exten => _123456,n,Dial(SIP/74,39,RcxX)
>> exten => _123456,n,Verbose(2,Voicemail for Main)
>> exten => _123456,n,Set(CALLERID(name)=)
>> exten => _123456,n,VoiceMail(74,us)
>> exten => _123456,n,Hangup
>> include => fax_incoming
>> include => michael_incoming
>> include => internal_calls
>>
>> exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
>> exten => h,n,Goto(noanswer,s,1)
>> exten => h,n(done),NoOp()
>>
>> Unfortunately two E-Mails are sent anyway...
>> This is the Asterisk log:
>>
>> -- Executing [00493529123456@michael_incoming:1]
>> Verbose("PJSIP/pbxmichael_in-0418", "2,Call for Main -
>> [+4935]") in new stack
>>   == Call for Main - [+4935]
>> -- Executing [00493529123456@michael_incoming:2]
>> Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX")
>> in new stack
>> -- Called local/123456@main_incoming
>> -- Executing [123456@main_incoming:1]
>> Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main -
>> [+4935]") in new stack
>>   == Call for Main - [+4935]
>> -- Executing [123456@main_incoming:2]
>> Set("Local/123456@main_incoming-0268;2",
>> "CALLERID(num)=035") in new stack
>> -- Executing [123456@main_incoming:3]
>> Set("Local/123456@main_incoming-0268;2",
>> "CHANNEL(musicclass)=default") in new stack
>> -- Executing [123456@main_incoming:4]
>> Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new
>> stack
>>   == Using SIP RTP CoS mark 5
>> -- Called SIP/74
>> -- Local/123456@main_incoming-0268;1 is ringing
>> -- SIP/74-0462 is ringing
>> -- Local/123456@main_incoming-0268;1 is ringing
>> -- SIP/74-0462 is ringing
>> -- SIP/74-0462 is ringing
>> -- SIP/74-0462 is ringing
>>   == Spawn extension (michael_incoming, 00493529123456, 2) exited
>> non-zero on 'PJSIP/pbxmichael_in-0418'
>> -- Executing [h@michael_incoming:1]
>> GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack
>> -- Executing [h@michael_incoming:2]
>> Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack
>> -- Goto (noanswer,s,1)
>>   == Spawn extension (main_incoming, 123456, 4) exited 

Re: [asterisk-users] Notifying missed calls

[Łukasz Grzywański]

Hi,
strange

  -- Goto (noanswer,s,1)
  -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418", "echo
"Verpasster Anruf vom +4935 um 19:13" | mail -s "Verpasster Anruf"
i...@mydomain.de") in new stack
  -- Executing [s@noanswer:1] NoOp("Local/123456@main_incoming-0268;2",
"UID CALL: 1636222382.6032/ DATE: 20211106-191306)") in new stack
  -- Executing [s@noanswer:2] System("Local/123456@main_incoming-0268;2",
"echo "Verpasster Anrufvom 035 um 19:13" | mail -s "Verpasster
Anruf" i...@mydomain.de") in new stack

pls run

asterisk -rx "dialplan show noanswer"

and please check:

[noanswer]
exten => s,1,NoOp(UID CALL: ${UNIQUEID} /
DATE:${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@.de)
exten => s,n,Hangup()


 LG
Lukasz

On Sat, 6 Nov 2021 at 19:20, Luca Bertoncello  wrote:

> Am 06.11.2021 um 15:06 schrieb Frank Vanoni:
>
> Hi Frank
>
> > The "h" extension is executed whenever a call is hang up in that
> > contexts.
> >
> > In your configuration it executes first the "s" extension (where you
> > GoTo h,1) and once that is executed, the "h" extension is executed
> > again.
>
> OK, I modified my configuration so:
>
> [main_incoming]
> exten => _00493529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _00493529123456,n,Dial(local/123456@main_incoming,,xX)
> exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _03529123456,n,Dial(local/123456@main_incoming,,xX)
> exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
> "+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
> exten => _123456,n,Set(CHANNEL(musicclass)=default)
> exten => _123456,n,Dial(SIP/74,39,RcxX)
> exten => _123456,n,Verbose(2,Voicemail for Main)
> exten => _123456,n,Set(CALLERID(name)=)
> exten => _123456,n,VoiceMail(74,us)
> exten => _123456,n,Hangup
> include => fax_incoming
> include => michael_incoming
> include => internal_calls
>
> exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
> exten => h,n,Goto(noanswer,s,1)
> exten => h,n(done),NoOp()
>
> Unfortunately two E-Mails are sent anyway...
> This is the Asterisk log:
>
> -- Executing [00493529123456@michael_incoming:1]
> Verbose("PJSIP/pbxmichael_in-0418", "2,Call for Main -
> [+4935]") in new stack
>   == Call for Main - [+4935]
> -- Executing [00493529123456@michael_incoming:2]
> Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX")
> in new stack
> -- Called local/123456@main_incoming
> -- Executing [123456@main_incoming:1]
> Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main -
> [+4935]") in new stack
>   == Call for Main - [+4935]
> -- Executing [123456@main_incoming:2]
> Set("Local/123456@main_incoming-0268;2",
> "CALLERID(num)=035") in new stack
> -- Executing [123456@main_incoming:3]
> Set("Local/123456@main_incoming-0268;2",
> "CHANNEL(musicclass)=default") in new stack
> -- Executing [123456@main_incoming:4]
> Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new
> stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/74
> -- Local/123456@main_incoming-0268;1 is ringing
> -- SIP/74-0462 is ringing
> -- Local/123456@main_incoming-0268;1 is ringing
> -- SIP/74-0462 is ringing
> -- SIP/74-0462 is ringing
> -- SIP/74-0462 is ringing
>   == Spawn extension (michael_incoming, 00493529123456, 2) exited
> non-zero on 'PJSIP/pbxmichael_in-0418'
> -- Executing [h@michael_incoming:1]
> GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack
> -- Executing [h@michael_incoming:2]
> Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack
> -- Goto (noanswer,s,1)
>   == Spawn extension (main_incoming, 123456, 4) exited non-zero on
> 'Local/123456@main_incoming-0268;2'
> -- Executing [h@main_incoming:1]
> GotoIf("Local/123456@main_incoming-0268;2", "0?done") in new stack
> -- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-0418",
> "UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack
> -- Executing [h@main_incoming:2]
> Goto("Local/123456@main_incoming-0268;2", "noanswer,s,1") in new stack
> -- Goto (noanswer,s,1)
> -- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418",
> "echo "Verpasster Anruf vom +4935 um 19:13" | mail -s
> "Verpasster Anruf" i...@mydomain.de") in new stack
> -- Executing [s@noanswer:1]
> NoOp("Local/123456@main_incoming-0268;2", "UID CALL: 1636222382.6032
> / DATE: 20211106-191306)") in new stack
> -- Executing [s@noanswer:2]
> System("Local/123456@main_incoming-0268;2", "echo "Verpasster Anruf
> vom 035 um 19:13" | mail -s "Verpasster Anruf"
> i...@mydomain.de") in new stack
>

Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 06.11.2021 um 15:06 schrieb Frank Vanoni:

Hi Frank

> The "h" extension is executed whenever a call is hang up in that
> contexts. 
> 
> In your configuration it executes first the "s" extension (where you
> GoTo h,1) and once that is executed, the "h" extension is executed
> again.

OK, I modified my configuration so:

[main_incoming]
exten => _00493529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _00493529123456,n,Dial(local/123456@main_incoming,,xX)
exten => _03529123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _03529123456,n,Dial(local/123456@main_incoming,,xX)
exten => _123456,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _123456,n,Set(CALLERID(num)=${IF($[ "${CALLERID(num):0:3}" =
"+49" ]?0${CALLERID(num):3}:${CALLERID(num)})})
exten => _123456,n,Set(CHANNEL(musicclass)=default)
exten => _123456,n,Dial(SIP/74,39,RcxX)
exten => _123456,n,Verbose(2,Voicemail for Main)
exten => _123456,n,Set(CALLERID(name)=)
exten => _123456,n,VoiceMail(74,us)
exten => _123456,n,Hangup
include => fax_incoming
include => michael_incoming
include => internal_calls

exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()

Unfortunately two E-Mails are sent anyway...
This is the Asterisk log:

-- Executing [00493529123456@michael_incoming:1]
Verbose("PJSIP/pbxmichael_in-0418", "2,Call for Main -
[+4935]") in new stack
  == Call for Main - [+4935]
-- Executing [00493529123456@michael_incoming:2]
Dial("PJSIP/pbxmichael_in-0418", "local/123456@main_incoming,,xX")
in new stack
-- Called local/123456@main_incoming
-- Executing [123456@main_incoming:1]
Verbose("Local/123456@main_incoming-0268;2", "2,Call for Main -
[+4935]") in new stack
  == Call for Main - [+4935]
-- Executing [123456@main_incoming:2]
Set("Local/123456@main_incoming-0268;2",
"CALLERID(num)=035") in new stack
-- Executing [123456@main_incoming:3]
Set("Local/123456@main_incoming-0268;2",
"CHANNEL(musicclass)=default") in new stack
-- Executing [123456@main_incoming:4]
Dial("Local/123456@main_incoming-0268;2", "SIP/74,39,RcxX") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/74
-- Local/123456@main_incoming-0268;1 is ringing
-- SIP/74-0462 is ringing
-- Local/123456@main_incoming-0268;1 is ringing
-- SIP/74-0462 is ringing
-- SIP/74-0462 is ringing
-- SIP/74-0462 is ringing
  == Spawn extension (michael_incoming, 00493529123456, 2) exited
non-zero on 'PJSIP/pbxmichael_in-0418'
-- Executing [h@michael_incoming:1]
GotoIf("PJSIP/pbxmichael_in-0418", "0?done") in new stack
-- Executing [h@michael_incoming:2]
Goto("PJSIP/pbxmichael_in-0418", "noanswer,s,1") in new stack
-- Goto (noanswer,s,1)
  == Spawn extension (main_incoming, 123456, 4) exited non-zero on
'Local/123456@main_incoming-0268;2'
-- Executing [h@main_incoming:1]
GotoIf("Local/123456@main_incoming-0268;2", "0?done") in new stack
-- Executing [s@noanswer:1] NoOp("PJSIP/pbxmichael_in-0418",
"UID CALL: 1636222382.6030 / DATE: 20211106-191306)") in new stack
-- Executing [h@main_incoming:2]
Goto("Local/123456@main_incoming-0268;2", "noanswer,s,1") in new stack
-- Goto (noanswer,s,1)
-- Executing [s@noanswer:2] System("PJSIP/pbxmichael_in-0418",
"echo "Verpasster Anruf vom +4935 um 19:13" | mail -s
"Verpasster Anruf" i...@mydomain.de") in new stack
-- Executing [s@noanswer:1]
NoOp("Local/123456@main_incoming-0268;2", "UID CALL: 1636222382.6032
/ DATE: 20211106-191306)") in new stack
-- Executing [s@noanswer:2]
System("Local/123456@main_incoming-0268;2", "echo "Verpasster Anruf
vom 035 um 19:13" | mail -s "Verpasster Anruf"
i...@mydomain.de") in new stack

Any other idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Notifying missed calls

[Frank Vanoni]

On Sat, 2021-11-06 at 14:46 +0100, Luca Bertoncello wrote:

> Really, I can't understand what you mean... I'm feeling really
> dumb...


No need to feel dumb. I'm not an expert and when I look to my
extensions.conf... well... countless pulling my hairs out, head banging
on the keyboard,,, :-)

The "h" extension is executed whenever a call is hang up in that
contexts. 

In your configuration it executes first the "s" extension (where you
GoTo h,1) and once that is executed, the "h" extension is executed
again.

Take a look to the example I posted. It's very basic, but it does the
job.




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Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 06.11.2021 um 14:43 schrieb Frank Vanoni:

Hi Frank

> On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote:
> 
>> 1) The E-Mails will be sent "double"
> 
> It sends the first mail by executing "noanswer,2" and a second mail
> because because of "main-incoming,h,2" 

Really, I can't understand what you mean... I'm feeling really dumb...

>> 2) The E-Mails will be sent for outgoing unanswered calls, too.
> 
> Use the "h" extension only in the context for incoming calls

I have just one "h" extension:

[main_incoming]
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()

Could you explain what you mean?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Notifying missed calls

[Frank Vanoni]

On Fri, 2021-11-05 at 10:50 +0100, Luca Bertoncello wrote:

> 1) The E-Mails will be sent "double"

It sends the first mail by executing "noanswer,2" and a second mail
because because of "main-incoming,h,2" 

> 2) The E-Mails will be sent for outgoing unanswered calls, too.

Use the "h" extension only in the context for incoming calls


> 



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Re: [asterisk-users] Notifying missed calls

[Frank Vanoni]

Here my configuration:

[incoming]
; Incoming from Swisscom
exten => +4191xxx,1,NoOp(Call from ${CALLERID(num)})
   same => n,Dial(SIP/deskphone,120)
   same => n,Hangup()

exten => h,1,GotoIf($["${DIALSTATUS}" = "ANSWER"]?done)
exten => h,n,System(echo "Missed Call from ${CALLERID(num)}" | mail -s
"Missed Call from ${CALLERID(num)}" my-em...@address.here)
exten => h,n(done),NoOp()





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Re: [asterisk-users] Notifying missed calls

[Łukasz Grzywański]

Hallo Luca,
Please show logs ( full )


On Fri, 5 Nov 2021 at 11:04, Luca Bertoncello  wrote:

> Am 03.11.2021 21:34, schrieb Antony Stone:
>
> Hi again
>
> > The n there should be 1, surely?
> >
> >> exten => h,n,Hangup
> >
> > I would say "remove that line".  The call has already been hung up, so
> > calling
> > Hangup is at best going to go into a recursive loop - it certainly
> > isn't going
> > to help.
>
> This is my current configuration:
>
> [cch]
> exten => _X.,1,Verbose(2,DEFAULT)
> include => internal_calls
> include => main_incoming
> include => fax_incoming
> include => michael_incoming
> include => myproxy
>
> [noanswer]
> exten => s,1,NoOp(UID CALL: ${UNIQUEID} / DATE:
> ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
> exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
> ${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@.de)
>
> 
>
> [main_incoming]
> exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
> exten => h,n,Goto(noanswer,s,1)
> exten => h,n(done),NoOp()
> exten => h,n,HangUp()
> ...
>
> It works, but I have two problems:
>
> 1) The E-Mails will be sent "double"
> 2) The E-Mails will be sent for outgoing unanswered calls, too.
>
> Do someone has an idea what is wrong in my configuration?
>
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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>
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Pozdrawiam,

Łukasz Grzywański
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Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 03.11.2021 21:34, schrieb Antony Stone:

Hi again


The n there should be 1, surely?


exten => h,n,Hangup


I would say "remove that line".  The call has already been hung up, so 
calling
Hangup is at best going to go into a recursive loop - it certainly 
isn't going

to help.


This is my current configuration:

[cch]
exten => _X.,1,Verbose(2,DEFAULT)
include => internal_calls
include => main_incoming
include => fax_incoming
include => michael_incoming
include => myproxy

[noanswer]
exten => s,1,NoOp(UID CALL: ${UNIQUEID} / DATE: 
${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um 
${STRFTIME(${EPOCH},,%H:%M)}" | mail -s "Verpasster Anruf" i...@.de)




[main_incoming]
exten => h,1,GotoIf($[“${DIALSTATUS}” = “ANSWER”]?done)
exten => h,n,Goto(noanswer,s,1)
exten => h,n(done),NoOp()
exten => h,n,HangUp()
...

It works, but I have two problems:

1) The E-Mails will be sent "double"
2) The E-Mails will be sent for outgoing unanswered calls, too.

Do someone has an idea what is wrong in my configuration?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Notifying missed calls

[Gerardo Barajas]

Hi Luca.

You can use the ${DIALSTATUS} channel variable.

https://www.voip-info.org/asterisk-variable-dialstatus/

GERARDO B

On Wed, Nov 3, 2021 at 2:52 PM Luca Bertoncello 
wrote:

> Hi list!
>
> I have this problem: I'd like to notify the user about missed calls.
> With "missed call" I mean: the caller calls, wait a while and hangup
> _before_ the voicemail starts.
>
> I got it call a script just before the voicemail starts, so:
>
> exten => s,1,Verbose(2,Call for Main - [${CALLERID(num)}])
> exten => s,n,Set(CHANNEL(musicclass)=default)
> exten => s,n,Dial(SIP/74,39,RcxX)
> exten => s,n,Verbose(2,Voicemail for Main)
> exten => s,n,Set(CALLERID(name)=)
> exten => s,n,Gosub(noanswer,s,1)
> exten => s,n,VoiceMail(74,us)
> exten => s,n,Hangup
>
> [noanswer]
> exten => s,1,NoOp(UID CALL: ${UNIQUEID} / DATE:
> ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
> exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
> ${STRFTIME(${EPOCH},,%H:%M)}" > /tmp/calllog.txt)
> exten => s,n,Return()
>
> but if the called hangs up prior the timeout for the voicemail, the
> Subrouting "noanswer" will not called...
>
> Any ideas?
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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>
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-- 
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--
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Proyectos Especiales/Preventa | www.neocenter.com
T:+52 (55)  8590-9000 x 7003
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Re: [asterisk-users] Notifying missed calls

[Carlos Rojas]

g: *When the called party hangs up*, continue to execute commands in the
current context at the next priority

On Wed, Nov 3, 2021 at 4:39 PM Luca Bertoncello 
wrote:

> Am 03.11.2021 um 21:34 schrieb Antony Stone:
> > On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:
> >
> >> I tried so:
> >>
> >> exten => h,n(hang),Gosub(noanswer,s,1)
> >
> > The n there should be 1, surely?
>
> Ach, you're right!
>
> Now it works!
>
> Thanks a lot
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 03.11.2021 um 21:34 schrieb Antony Stone:
> On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:
> 
>> I tried so:
>>
>> exten => h,n(hang),Gosub(noanswer,s,1)
> 
> The n there should be 1, surely?

Ach, you're right!

Now it works!

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Notifying missed calls

[Antony Stone]

On Wednesday 03 November 2021 at 21:29:46, Luca Bertoncello wrote:

> I tried so:
> 
> exten => h,n(hang),Gosub(noanswer,s,1)

The n there should be 1, surely?

> exten => h,n,Hangup

I would say "remove that line".  The call has already been hung up, so calling 
Hangup is at best going to go into a recursive loop - it certainly isn't going 
to help.


Antony.

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 please *don't* CC me.

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Re: [asterisk-users] Notifying missed calls

[Luca Bertoncello]

Am 03.11.2021 um 21:24 schrieb Doug Lytle:
 but if the called hangs up prior the timeout for the voicemail, the
 Subrouting "noanswer" will not called...
> 
> You can use the h priority for that.
> 
> https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions

Hi Doug,

could you send me an example?

I tried so:


exten => h,n(hang),Gosub(noanswer,s,1)
exten => h,n,Hangup
exten => _xx,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => _xx,n,Set(CHANNEL(musicclass)=default)
exten => _xx,n,Dial(SIP/74,39,RcxX)
exten => _xx,n,Verbose(2,Voicemail for Main)
exten => _xx,n,Set(CALLERID(name)=)
exten => _xx,n,Gosub(noanswer,s,1)
exten => _xx,n,VoiceMail(74,us)
exten => _xx,n,Hangup

But it does not work...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Notifying missed calls

[Doug Lytle]

>>> but if the called hangs up prior the timeout for the voicemail, the
>>> Subrouting "noanswer" will not called...

You can use the h priority for that.

https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions

Doug

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[asterisk-users] Notifying missed calls

[Luca Bertoncello]

Hi list!

I have this problem: I'd like to notify the user about missed calls.
With "missed call" I mean: the caller calls, wait a while and hangup
_before_ the voicemail starts.

I got it call a script just before the voicemail starts, so:

exten => s,1,Verbose(2,Call for Main - [${CALLERID(num)}])
exten => s,n,Set(CHANNEL(musicclass)=default)
exten => s,n,Dial(SIP/74,39,RcxX)
exten => s,n,Verbose(2,Voicemail for Main)
exten => s,n,Set(CALLERID(name)=)
exten => s,n,Gosub(noanswer,s,1)
exten => s,n,VoiceMail(74,us)
exten => s,n,Hangup

[noanswer]
exten => s,1,NoOp(UID CALL: ${UNIQUEID} / DATE:
${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}))
exten => s,n,System(echo "Verpasster Anruf vom ${CALLERID(NUM)} um
${STRFTIME(${EPOCH},,%H:%M)}" > /tmp/calllog.txt)
exten => s,n,Return()

but if the called hangs up prior the timeout for the voicemail, the
Subrouting "noanswer" will not called...

Any ideas?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Block Spam Calls

[Alexander Perkins]

Hi All.  Thank you so much for all the feedback; it is really helpful.  I
found a company out there that specializes in Asterisk solutions and has a
Robocall/SPAM call solution specifically for Asterisk.  We give it a spin
last week and it turned out great.  It's not free, but works very well and
I think it is worth passing it on.  Here's the site for the service -
https://www.tiltx.com/asterisk-robocall-blocker.

I thought I'd pass this along as it has taken me some time to find
something for Asterisk that actually works.

Hope this helps,
Alex
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Re: [asterisk-users] Block Spam Calls

[Julian Beach]

Hello Jeff,

Friday, December 13, 2019, 7:42:38 PM, you wrote:

> Mind posting your dialplan code?  I was thinking the same thing - very
> much like an old spam control program I used to use whose name now
> escapes me.  First time senders would have to respond to an auto-reply,
> then were added to a whitelist.

This  is not the neatest of code segments, as there are a few checks I
make before I let calls through to the phones:

(1) It  checks  for  blacklisted  callers, and sends them off to a go-away
message

(2) It checks whether there is Caller ID information, and sends sends
anon calls to another handler (again asking callers to press 1 or
leave a message) but there is also a switch to just send them to
voicemail: I had a problem with 1 anon caller phoning repeatedly a few
years ago, and that was the solution!

(3) For calls with a CID number, it checks whether these have called
before and if not, adds them to the previous caller db then sends them
to an IVR to play them a message and ask them to press a key (this
means that if they call again, they will always go through to the phones, as
a  number  of  genuine callers cannot, or won't, dial 1 - if they turn
out to be a robocall, they are blacklisted)

(4) Calls that pass all the tests go through to the phones

I  am  sure that I could make the code much more efficient, but it has
not  been  a  priority and I have been adding features as I need them,
rather than re-writing!


[from-pstn] (initial handler for PSTN calls)
exten => s,1,Log(NOTICE, Incoming PSTN Call from CDR ${CDR(src)} and CID 
${CALLERID(name)})
exten => s,n,GotoIf(${DB_EXISTS(blacklist/${CDR(src)})}?block) ; Check whether 
caller blacklisted
exten => s,n,Set(DB(callers/lastcall)=${CDR(src)})
exten => s,n,Set(GLOBAL(CALLEDLINE)=L1-)
exten => s,n,GotoIf($["${CDR(src)}" != ""]?toincoming) ; if there is valid CID 
go to IncomingHome
exten => s,n,GotoIf($["${CALLERID(name)}" = "WITHHELD"]?nocid) ; if anon call...
exten => s,n,GotoIf($["${CALLERID(name)}" = "INTERNATIONAL"]?nocid)
exten => s,n,GotoIf($["${CALLERID(name)}" = "UNAVAILABLE"]?nocid)
exten => s,n,GotoIf($["${CALLERID(name)}" = "PAYPHONE"]?nocid)
exten => s,n,GotoIf($["${CDR(src)}" = ""]?drop)
exten => s,n(toincoming),GotoIf(${DB_EXISTS(previous/${CDR(src)})}?:unknown) ; 
Check whether caller has rung before
exten => s,n,GoSub(IncomingHome,s,1) ; answer the call in the macro
exten => s,n(nocid),GotoIf($[ $[ "${DB(variables/ANONTOVM)}" = "1" ] ]?vm) ; 
...check whether anon goes straight to vm
exten => s,n,GoSub(IncomingHome,s,1) ; answer the call in the macro
exten => s,n(block),GoSub(Handler-MarketingCall,s,1) ; deal with blacklisted 
callers
exten => s,n(vm),Log(NOTICE, No CID data or AnontoVM set) ; send anon callers 
to AnonCall IVR
exten => s,n,Goto(AnonCall-menu,s,1)
exten => s,n(unknown),Log(NOTICE, Unknown caller to IVR) ; send unknown callers 
to UnknownCaller IVR
exten => s,n,Set(DB(previous/${CDR(src)})=1) ; add number to previous caller DB
exten => s,n,GoSub(UnknownCaller-menu,s,1,(${CDR(src)}))
exten => s,n(drop),Log(NOTICE, No CID data, call dropped) ; drop calls with no 
CID info
exten => s,n,Wait(15)
exten => s,n,Hangup()


[UnknownCaller-menu]  ; The  Unknown  caller IVR - with some logging of
call volumes

exten => s,1,Log(NOTICE,Unknown Caller ${ARG1} - Caller to IVR)
exten => s,n,Verbose(2,Anon Calls = ${DB(unknown/total)} Total, 
${DB(unknown/valid)} Valid, ${DB(unknown/invalid)} Invalid)
exten => s,n,Set(unknown=$[${DB(unknown/total)} + 1])
exten => s,n,Set(DB(unknown/total)=${unknown})
exten => s,n,Verbose(2,Unknown Calls set to ${DB(unknown/total)})
exten => s,n,Answer(500)
exten => s,n(loop),Background(unknown)
exten => s,n,WaitExten()
; 1 - Call to incoming call handler
exten => 1,1,NoOp(Real caller progressing)
exten => 1,n,Log(NOTICE,Unknown Call - Caller dialled to continue to phones)
exten => 1,n,Set(valid=$[${DB(unknown/valid)} + 1])
exten => 1,n,GoSub(IncomingHome,s,1)
exten => 1,n,Hangup()
; 2 - invalid entry
exten => i,1,Playback(invalid)
exten => i,n,Log(NOTICE,Unknown Call - Caller dialled invalid number)
exten => i,n,Set(invalids=$[${DB(unknown/invalid)} + 1])
exten => i,n,Set(DB(unknown/invalid)=${invalids})
exten => i,n,Goto(s,loop)
; 3 - timeout
exten => t,1,Playback(invalid)
exten => t,n,Log(NOTICE,Unknown Call - Caller timed out)
exten => t,n,Set(invalids=$[${DB(unknown/invalid)} + 1])
exten => t,n,Set(DB(unknown/invalid)=${invalids})
exten => t,n,GoSub(subMessaging,s,1,(${ARG1}),("Unknown"))
exten => t,n,Hangup()



--
Best regards,
 Julianmailto:jb_s...@trink.co.uk 


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Re: [asterisk-users] Block Spam Calls

[Doug Lytle]

On 12/13/19 11:48 AM, Julian Beach wrote:

Hello Doug,

Friday, December 13, 2019, 11:03:37 AM, you wrote:


This is exactly what I do - “press 1 for a human”
Works great

I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call directly on without
the IVR prompts.



For those that would like to see my code:

exten => 517xxx,1,Answer()
    same => n,Gosub(check_blacklist,s,1)
    same => n,Gosub(get_callerid,s,1)
    same => n,Gosub(check_for_direct,s,1)
    same => n,Set(CHANNEL(musicclass)=music)
    same => n,Gosub(extension_timeouts,s,1)
    same => n,Dial(SIP/3501,${timeout.timeout},TtKk)
    same => n,NoOP(Dial Status: ${DIALSTATUS})
    same => n,NoOP(Hangup Cause: ${HANGUPCAUSE})
    same => n,Gosub(s-${DIALSTATUS},s,1)

[check_for_direct]

;**
;* Check if there is a match of the inbound call to the speed dial list
;* If not, make then go through the IVR menu
;***

exten => 
s,1,Set(ARRAY(speed.phone,speed.name)=${ODBC_MENU_DIRECT(drdos,${CALLERID(number)})})


;
;* If the contents of speed.phone is blank, assume that it
;* is not programmed and force the call to use the IVR to
;* prove they are not an automated call.
;

 same => n,GotoIf($["${speed.phone}" != "" ]?3:ivr_menu,s,1)
 same => n,NoOP(${speed.name} is on the approved list)
 same => n,Return()
 same => n,Hangup()



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Re: [asterisk-users] Block Spam Calls

[Jeff LaCoursiere]

Mind posting your dialplan code?  I was thinking the same thing - very 
much like an old spam control program I used to use whose name now 
escapes me.  First time senders would have to respond to an auto-reply, 
then were added to a whitelist.


This would be a great FreeBSD module...

Cheers,

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* 
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

 
 



On 12/13/19 10:48 AM, Julian Beach wrote:

Hello Doug,

Friday, December 13, 2019, 11:03:37 AM, you wrote:


This is exactly what I do - “press 1 for a human”
Works great

I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call directly on without
the IVR prompts.

I do something similar for calls without caller ID, but I was still
getting robocalls with spoofed caller ID. I have now changed the dialplan
slightly so that the first time people call they are asked to dial 1.
After the first call, they are added to a known caller list and get
straight through, and any robocalls at that point are blacklisted
manually. I have found that most robocallers spoof the Caller ID so
rarely call from the same number twice. It means that legitimate
callers who cannot dial 1 just have to dial again to get through to
the phones - there is a recorded message telling them to dial 1 or
call back. I haven't had a robocall since!

The hardest thing about this was extracting all the numbers of
previous callers from the CDR and adding it to the Previous_Callers
AstDB for the lookup. I didn't want to make existing callers go through
the initial learning process.

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Re: [asterisk-users] Block Spam Calls

[Julian Beach]

Hello Doug,

Friday, December 13, 2019, 11:03:37 AM, you wrote:

>> This is exactly what I do - “press 1 for a human”
>> Works great

> I do this as well, but I also do a database lookup to see if the number
> is on our speeddial list and if so, pass the call directly on without 
> the IVR prompts.

I do something similar for calls without caller ID, but I was still
getting robocalls with spoofed caller ID. I have now changed the dialplan
slightly so that the first time people call they are asked to dial 1.
After the first call, they are added to a known caller list and get
straight through, and any robocalls at that point are blacklisted
manually. I have found that most robocallers spoof the Caller ID so
rarely call from the same number twice. It means that legitimate
callers who cannot dial 1 just have to dial again to get through to
the phones - there is a recorded message telling them to dial 1 or
call back. I haven't had a robocall since!

The hardest thing about this was extracting all the numbers of
previous callers from the CDR and adding it to the Previous_Callers
AstDB for the lookup. I didn't want to make existing callers go through
the initial learning process.

-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk


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Re: [asterisk-users] Block Spam Calls

[Doug Lytle]

On 12/12/19 6:55 PM, Adam Goldberg wrote:

This is exactly what I do - “press 1 for a human”
Works great


I do this as well, but I also do a database lookup to see if the number 
is on our speeddial list and if so, pass the call directly on without 
the IVR prompts.


Doug


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Re: [asterisk-users] Block Spam Calls

[Greg Troxel]

D'Arcy Cain  writes:

> Not bad.  I was toying with another idea.  I find that if I don't answer
> a robot fast enough it just hangs up.  How about ring two or three times
> before passing to the actual extension?

You could try that and let us know, but I suspect:

  some robocallers don't hang up so fast and will get through

  this will annoy the legit callers

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Re: [asterisk-users] Block Spam Calls

[D'Arcy Cain]

On 12/12/19 5:33 PM, Greg Woods wrote:
> Most spam calls are robocalls these days. At my house, I can block
> pretty much all of the robocalls by requiring the caller to take some
> action before ringing the phones. In our case, the action is just to
> dial 1 for my wife or 2 for me. The only difference it makes in the end
> is which voice mailbox the caller gets transferred to if no one answers,
> but since asterisk so programmable, there are a lot of ways to
> accomplish the basic concept. I can see tons of calls in my log
> that never get through to bother us.

Not bad.  I was toying with another idea.  I find that if I don't answer
a robot fast enough it just hangs up.  How about ring two or three times
before passing to the actual extension?

-- 
D'Arcy J.M. Cain
Vybe Networks Inc.
A unit of Excelsior Solutions Corporation - Propelling Business Forward
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

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Re: [asterisk-users] Block Spam Calls

[Adam Goldberg]

This is exactly what I do - “press 1 for a human”
Works great


From: asterisk-users  on behalf of 
Greg Woods 
Sent: Thursday, December 12, 2019 6:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Block Spam Calls

Most spam calls are robocalls these days. At my house, I can block pretty much 
all of the robocalls by requiring the caller to take some action before ringing 
the phones. In our case, the action is just to dial 1 for my wife or 2 for me. 
The only difference it makes in the end is which voice mailbox the caller gets 
transferred to if no one answers, but since asterisk so programmable, there are 
a lot of ways to accomplish the basic concept. I can see tons of calls in my 
log that never get through to bother us.

--Greg


On Tue, Dec 10, 2019 at 8:58 AM Alexander Perkins 
mailto:alexanderhenryperk...@gmail.com>> wrote:
Hi All.  Does anybody know if Google/Android has an API I can sign up for that 
will allow us to query the caller ID and find out if it is spam or a 
robocaller?  I ask because we've had increase in spam calls and I'd like to 
simply play dead air or something really annoying.

Thanks all,
Alex
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Re: [asterisk-users] Block Spam Calls

[Greg Woods]

Most spam calls are robocalls these days. At my house, I can block pretty
much all of the robocalls by requiring the caller to take some action
before ringing the phones. In our case, the action is just to dial 1 for my
wife or 2 for me. The only difference it makes in the end is which
voice mailbox the caller gets transferred to if no one answers, but since
asterisk so programmable, there are a lot of ways to accomplish the basic
concept. I can see tons of calls in my log that never get through to bother
us.

--Greg


On Tue, Dec 10, 2019 at 8:58 AM Alexander Perkins <
alexanderhenryperk...@gmail.com> wrote:

> Hi All.  Does anybody know if Google/Android has an API I can sign up for
> that will allow us to query the caller ID and find out if it is spam or a
> robocaller?  I ask because we've had increase in spam calls and I'd like to
> simply play dead air or something really annoying.
>
> Thanks all,
> Alex
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Re: [asterisk-users] Block Spam Calls

[Ira]

Title: Re: [asterisk-users] Block Spam Calls


Hello Alexander,

Tuesday, December 10, 2019, 7:57:54 AM, you wrote:





Hi All.  Does anybody know if Google/Android has an API I can sign up for that will allow us to query the caller ID and find out if it is spam or a robocaller?  I ask because we've had increase in spam calls and I'd like to simply play dead air or something really annoying.



I've had reasonable luck making a list of suspect Caller IDs though they seem to be slowing down the use of those as time goes by.  In my experience, calls with these caller IDs are never valid calls. Depending on who's calling you may also have good luck blocking all calls with the same area code and 3 digit prefix. I store all this in the Asterisk database and wrote my own blacklist and whitelist functions to handle it in the dial plan. If you do this you'll need a whitelist to let the people who end up here because they've not published their caller ID to the lookup lists and one of get the default "800 Service" tags.

Ira


000                                
                              
0                              
00                            
000                            
800 SERVICE                            
800 Service                            
800_SERVICE                            
800_Service                            
ASSIST                                
CLIENT SUPPORT                        
COMMERCIAL RECO                        
CUST SERVICE                          
CUST SRVC                              
CUSTOMER SERVI                        
CUSTOMER SERVIC                        
CUSTOMER SERVICE                      
CUSTOMER SRV                          
CUSTOMER-SERVI                        
CUSTOMER-SERVIC                        
CUSTOMER-SERVICE                      
CUSTOMER_SERVI                        
CUSTOMER_SERVIC                        
CUSTOMER_SERVICE                      
INS SRVCS                              
INSURANCE                              
INSURANCE SRVC                        
LOCAL RANKING                          
PRIORITY MARKET                        
PRIVATE NAME                          
RENEWAL DEPT                          
RESEARCH CENTER                        
TELEMARKETER                          
TOLL FREE                              
TOLL FREE CALL                        
TOLL-FREE                              
TOLL-FREE  -                          
TOLL-FREE -                            
TOLL-FREE-                            
TOLL_FREE                              
TV RESEARCH                            
TV RESEARCH CO                        
VOIP CALLER                            





-- Ira


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Re: [asterisk-users] Block Spam Calls

[Hermann Wecke]

On Tue, Dec 10, 2019 at 11:01 AM Alexander Perkins
 wrote:
> Hi All.  Does anybody know if Google/Android has an API I can sign up for 
> that will allow us to query the caller ID and find out if it is spam or a 
> robocaller?

I don't think that there is a public (free) API. All robocall
protection services are paid ones. You can find several on Twilio from
US$ 0.003 up to US$ 0.06 per query, depending on what you are looking
for.

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[asterisk-users] Block Spam Calls

[Alexander Perkins]

Hi All.  Does anybody know if Google/Android has an API I can sign up for
that will allow us to query the caller ID and find out if it is spam or a
robocaller?  I ask because we've had increase in spam calls and I'd like to
simply play dead air or something really annoying.

Thanks all,
Alex
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[asterisk-users] Check other calls on same endpoint (validate / screen customer supplied Diversion / From header)

[Benoit Panizzon]

Hi Gang

Yes, big project on the rise to do things better / more flexible than
our existing commercial TSP switch.

During call screening process, we would like to allow customers to send
the original callingID in a attended call diversion scenario.

From the Voice Switch point of view, there are two call legs involved.
So on one call leg, I would like to check the callerID on the second
call leg. Is this doable?

Example:

Alice calls Bob who does picks up the call and puts it throught to
Charlie. Charlie shall see Alices CallerID:

Call Leg 1:
===
Invite: Bob
From: Alice
PAI: Bob

Call Leg 2:
===
Invite: Charlie
From: Alice
PAI: Bob
Diversion: Bob, Reason: Attended-Transfer

The screening on Leg 2 would normally not allow Alice to be sent as
Caller.
But if the sceening on Leg 2 would be able to look up Leg 1 and see the
co-existing call From: Alice To: Bob, it could allow Alice to be sent
out (and maybe add a missing Diversion and/or PAI header or also screen
the Diversion header).

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
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__

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[asterisk-users] Problems with calls dropping on Android.

[Sebastian Nielsen]

Hello.

I have the following in sip.conf

[sip09]

type=peer

defaultuser=sip09

nat=yes

qualify=no

secret=sip09

host=dynamic

context=outgoing

dtmfmode=rfc2833

disallow=all

allow=ulaw

allow=alaw

allow=h263p

deny=0.0.0.0/0.0.0.0

permit=192.168.2.2/255.255.255.255

jbenable = yes

jbforce = yes

jbmaxsize = 100

jbresyncthreshold = 200

jbimpl = fixed

transport=tcp

sendrpid=yes

 

And these settings in Android native client.

 

Username: sip09

Password: sip09

Server: 192.168.1.10

Username at authentication: sip09

Display name: Same as username

Outgoing proxy: 192.168.1.10

Port: 5060

Transport: TCP

Send keep alive: Always

 

However, if I make a call FROM android phone, call is dropped after 30
seconds, regardless of answer or not. If I make call TO android phone, it
works normally.

No NAT problems inbetween, there is a VPN between the phone and SIP server
with full access.

 

I guess I need to do some trick to have it work with Android. Apparently the
packets are received in both ends - else audio wouldn't work, but guess the
stock native SIP client on android ignores certain packets right?

This is an Android 9 phone.

 

 

Additionally, I wonder if its possible to change the callerid shown in
display when calling out? Like RPID. It works on my desktop phones, if I
enter a short code, the full name and number is shown on display, but on the
Android phone, it doesn't work, only the dialled shortnumber is shown.

Also I wonder if its possible to have asterisk send the remote callerid
(when receiving a call) in such a way it gets stored in call log with full
names and such - without having to resort to using phonebook.

 

 

SIP debug log:

 

*CLI> sip set debug ip 192.168.2.2

SIP Debugging Enabled for IP: 192.168.2.2

*CLI> Really destroying SIP dialog
'6f9956035553ab1b79ca057f5dffe0ac@192.168.2.2' Method: OPTIONS

Really destroying SIP dialog 'fc3307059c816094a6c6ce100cf383e5@192.168.2.2'
Method: OPTIONS

 

<--- SIP read from TCP:192.168.2.2:51729 --->

OPTIONS sip:192.168.1.10 SIP/2.0

Call-ID: e65234cb818a143bc3c167a782b98e96@192.168.2.2

CSeq: 3984 OPTIONS

From: "sip09" ;tag=3997716169

To: "sip09" 

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;rport

Max-Forwards: 70

User-Agent: SIPAUA/0.1.001

Content-Length: 0

 

<->

--- (9 headers 0 lines) ---

Sending to 192.168.2.2:51729 (no NAT)

Looking for s in cellip (domain 192.168.1.10)

 

<--- Transmitting (no NAT) to 192.168.2.2:51729 --->

SIP/2.0 200 OK

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bK105b3648c13a72f8fbe7ce3049df71aa3130;receive
d=192.168.2.2;rport=51729

From: "sip09" ;tag=3997716169

To: "sip09" ;tag=as4c9bb00e

Call-ID: e65234cb818a143bc3c167a782b98e96@192.168.2.2

CSeq: 3984 OPTIONS

Server: Asterisk PBX 13.21.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

Supported: replaces,timer

Contact: 

Accept: application/sdp

Content-Length: 0

 

 

<>

Scheduling destruction of SIP dialog
'e65234cb818a143bc3c167a782b98e96@192.168.2.2' in 32000 ms (Method: OPTIONS)

 

<--- SIP read from TCP:192.168.2.2:51729 --->

INVITE sip:02@192.168.1.10 SIP/2.0

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

CSeq: 9116 INVITE

From: "sip09" ;tag=3432177901

To: 

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;rport

Max-Forwards: 70

Contact: "sip09" 

Content-Type: application/sdp

Content-Length: 295

 

v=0

o=- 1571035683065 1571035683066 IN IP4 192.168.2.2

s=-

c=IN IP4 192.168.2.2

t=0 0

m=audio 26726 RTP/AVP 96 97 3 0 8 127

a=rtpmap:96 GSM-EFR/8000

a=rtpmap:97 AMR/8000

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-15

<->

--- (10 headers 13 lines) ---

Sending to 192.168.2.2:51729 (no NAT)

Sending to 192.168.2.2:51729 (no NAT)

Using INVITE request as basis request -
fcaad738faee2d0250d0cf2366139979@192.168.2.2

Found peer 'sip09' for 'sip09' from 192.168.2.2:51729

 

<--- Reliably Transmitting (NAT) to 192.168.2.2:51729 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/TCP
192.168.2.2:56334;branch=z9hG4bKf8bf8138c906000bc3f8601a5df558943130;receive
d=192.168.2.2;rport=51729

From: "sip09" ;tag=3432177901

To: ;tag=as4d53b5f5

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

CSeq: 9116 INVITE

Server: Asterisk PBX 13.21.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

Supported: replaces,timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dc98e50"

Content-Length: 0

 

 

<>

Scheduling destruction of SIP dialog
'fcaad738faee2d0250d0cf2366139979@192.168.2.2' in 32000 ms (Method: INVITE)

 

<--- SIP read from TCP:192.168.2.2:51729 --->

ACK sip:02@192.168.1.10 SIP/2.0

Call-ID: fcaad738faee2d0250d0cf2366139979@192.168.2.2

Max-Forwards: 70

From: "sip09" ;tag=3432177901

To: 

Re: [asterisk-users] AMI mulitple calls quickly

[Jerry Geis]

Thanks all for the suggestions.

1) ActionID. I was not using, so that could definitely have helped.
2) I was using Async already.
3) I changed to open multiple AMI connections - Worked like a champ.
4) Used to use call files but slow to first action... So changed to AMI
long time ago.

Thanks,

Jerry

>
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Re: [asterisk-users] AMI mulitple calls quickly

[Carlos Chavez]

On 3/12/19 11:03 AM, Steve Edwards wrote:


On Mon, 11 Mar 2019, Jerry Geis wrote:

If I use the AMI interface to originate a call, close the connection, 
open another connection etc...This works. but is slow...


Would opening multiple AMI connections be an option?

    You should be able to send multiple originate commands without 
having to wait as long as you are using the "Async" option in your 
originate action.  That way you just send all your calls and then wait 
for an "OriginateResponse" event to know the result of each of your 
calls.  Just remember to also use an "ActionID" to identify each call so 
you can match the response event.


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Carlos Chávez
+52 (55)8116-9161


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Re: [asterisk-users] AMI mulitple calls quickly

[Steve Edwards]

On Mon, 11 Mar 2019, Jerry Geis wrote:

If I use the AMI interface to originate a call, close the connection, 
open another connection etc...This works. but is slow...


Would opening multiple AMI connections be an option?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] AMI mulitple calls quickly

[Antony Stone]

On Tuesday 12 March 2019 at 01:19:37, Jerry Geis wrote:

> Lets say I have to make 40 phone calls quickly.
> 
> If I use the AMI interface to originate a call, close the connection, open
> another connection etc...
> This works. but is slow...

How about using call files instead?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files

https://www.voip-info.org/asterisk-auto-dial-out/

I think that may work better for you.


Antony.

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[asterisk-users] AMI mulitple calls quickly

[Jerry Geis]

Lets say I have to make 40 phone calls quickly.

If I use the AMI interface to originate a call, close the connection, open
another connection etc...
This works. but is slow...

If I open the AMI interface and originate a call - DONT close the interface
, get the response, originate another call, some of the calls are missed.
even though I get OK response.
(All calls are Async mode).

If I open the AMI interface and orignate a call - DONT close the interface
but sleep 1 second (again slow). this seems to work.

Question is - how do I open the AMI, send the originate command, get the
valid response and send the next originate command and have it actually do
it.  How is that done ?

I am using asterisk 13.25.0,
Thanks,

Jerry
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[asterisk-users] Problem receiving calls with Telmex in Mexico...

[Carlos Chavez]

    Hi.  I am having a problem when trying to receive calls via en E1 
from  Telmex using MFC/R2 (MX Variant).  Outgoing calls are fine.  We 
are using a PBXact system with a Digium TE420 (5th Gen) card.  Here is a 
log from the call:


[10:46:37:707] [Thread: 140631230322432] [Chan 1] - Call started at Sat 
Jan 12 10:46:37 2019 on chan 1 [openr2 version 1.3.3, revision (release)]
[10:46:37:707] [Thread: 140631230322432] [Chan 1] - Initialized R2 MF 
detector
[10:46:37:707] [Thread: 140631230322432] [Chan 1] - CAS Tx >> [SEIZE 
ACK] 0x0C

[10:46:37:707] [Thread: 140631230322432] [Chan 1] - CAS Raw Tx >> 0x0D
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - MF Rx << 2 [ON]
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 0

[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Cannot cancel timer 0
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 2
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - DNIS so far: 2, 
expected length: 4
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:37:781] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:37:781] [Thread: 140631230322432] [Chan 1] - scheduled timer id 2 
(mf_back_cycle)

[10:46:37:861] [Thread: 140631230322432] [Chan 1] - MF Rx << 2 [OFF]
[10:46:37:861] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - MF Rx << 8 [ON]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 2
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - timer id 2 found, 
cancelling it now

[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 8
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - DNIS so far: 28, 
expected length: 4
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:37:921] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:37:921] [Thread: 140631230322432] [Chan 1] - scheduled timer id 3 
(mf_back_cycle)

[10:46:38:001] [Thread: 140631230322432] [Chan 1] - MF Rx << 8 [OFF]
[10:46:38:001] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [ON]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 3
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - timer id 3 found, 
cancelling it now

[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 0
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - DNIS so far: 280, 
expected length: 4
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - Requesting next DNIS 
with signal 0x31.

[10:46:38:061] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [ON]
[10:46:38:061] [Thread: 140631230322432] [Chan 1] - scheduled timer id 4 
(mf_back_cycle)

[10:46:38:121] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [OFF]
[10:46:38:121] [Thread: 140631230322432] [Chan 1] - MF Tx >> 1 [OFF]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [ON]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 4
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - timer id 4 found, 
cancelling it now

[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Getting DNIS digit 0
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - DNIS so far: 2800, 
expected length: 4

[10:46:38:181] [Thread: 140631230322432] [Chan 1] - Done getting DNIS!
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - MF Tx >> 6 [ON]
[10:46:38:181] [Thread: 140631230322432] [Chan 1] - scheduled timer id 5 
(mf_back_cycle)

[10:46:38:261] [Thread: 140631230322432] [Chan 1] - MF Rx << 0 [OFF]
[10:46:38:261] [Thread: 140631230322432] [Chan 1] - MF Tx >> 6 [OFF]
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Bits changed from 
0x00 to 0x08
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - CAS Rx << [CLEAR 
FORWARD] 0x08
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Far end 
disconnected. Reason: Normal Clearing

[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Call ended
[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Attempting to cancel 
timer timer 0

[10:46:38:577] [Thread: 140631230322432] [Chan 1] - Cannot cancel timer 0

Basically all I see on the CLI is that a call comes in to Channel 1 and 
hangs up immediately.  Here is the system.conf:


span=1,1,0,CAS,HDB3
span=2,1,0,CAS,HDB3
span=3,1,0,CAS,HDB3
span=4,1,0,CAS,HDB3
cas=1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124:1101
echocanceller=oslec,1-15,17-31,32-46,48-62,63-77,79-93,94-108,110-124
loadzone=mx

Here is the relevant chan_dahdi.conf (it is auto generated by PBXact so 
it is in several files:


[general]

; generated by module
#include chan_dahdi_general.conf

; for user additions not provided by module
#include chan_dahdi_general_custom.conf

[channels]
language=es
busydetect=yes
busycount=5
usecallerid=no
callwaiting=no
usecallingpres=no
threewaycalling=no
transfer=yes

Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Joshua Colp]

On Fri, Jun 9, 2017, at 03:44 PM, Michael Maier wrote:



> 
> Further investigation showed, that Telekom provides the line info in the
> Request Line (as seen by Wireshark):
> 
> Request-Line: INVITE sip:+49@46.37.15.4:5060;line=azpreyb SIP/2.0
> 
> You can't find it if you expect it in contact header - or do you expect
> it in the Request-Line?

Recent code checks the To URI and the Request URI.

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Michael Maier]

On 06/09/2017 at 08:44 PM Michael Maier wrote:
> On 06/08/2017 at 10:22 PM Michael Maier wrote:
>> Hello Joshua,
>>
>> thank you very much for your extremely quick answer! I really appreciate
>> your work and your friendly and your patient support!
>>
>>
>> On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
>>> On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
 Hello!

 I've got a problem to select the correct trunk if there is one provider
 and different numbers with different configurations for this same
 provider.

 Example:

 trunk-prov1-2345
 trunk-prov1-2346
 trunk-prov1-2347

 Each trunk registers an own number (at the same provider) and provides
 own configuration: they have different allowed codecs e.g..

 What I'm experiencing now, is, that each incoming call is provided by
 trunk-prov1-2346, no matter which number has been dialed.

 The problem isn't the routing (this is done on base of the correct DID),
 but the problem is, that wrong codices are used if the wrong trunk is
 selected.

 Is this a problem of asterisk or is it caused by the provider, which
 always addresses the same "trunk" regardless which number has been
 called?
>>>
>>> Asterisk is the one who associates an incoming message with an endpoint.
>>> In the case of providers you can use IP based matching - which would
>>> behave as you see, only one can be matched. The second option is the
>>> line option[1] which may or may not work (it depends on the behavior of
>>> the provider). If it works then the right endpoint would be chosen. Out
>>> of those two options there's nothing else applicable built in to match.
>>>
>>> [1]
>>> http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
>>>
>>
>> Unfortunately Deutsche Telekom doesn't support this solution :-(.
> 
> Further investigation showed, that Telekom provides the line info in the
> Request Line (as seen by Wireshark):
> 
> Request-Line: INVITE sip:+49@46.37.15.4:5060;line=azpreyb SIP/2.0
> 
> You can't find it if you expect it in contact header - or do you expect
> it in the Request-Line?

Ok - got it.

It's necessary, that the value given for endpoint= is exactly the same
name as used for the trunk name itself and the match option for this
trunk should be omitted completely.


Thanks,
Michael

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Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Michael Maier]

On 06/08/2017 at 10:22 PM Michael Maier wrote:
> Hello Joshua,
> 
> thank you very much for your extremely quick answer! I really appreciate
> your work and your friendly and your patient support!
> 
> 
> On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
>> On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
>>> Hello!
>>>
>>> I've got a problem to select the correct trunk if there is one provider
>>> and different numbers with different configurations for this same
>>> provider.
>>>
>>> Example:
>>>
>>> trunk-prov1-2345
>>> trunk-prov1-2346
>>> trunk-prov1-2347
>>>
>>> Each trunk registers an own number (at the same provider) and provides
>>> own configuration: they have different allowed codecs e.g..
>>>
>>> What I'm experiencing now, is, that each incoming call is provided by
>>> trunk-prov1-2346, no matter which number has been dialed.
>>>
>>> The problem isn't the routing (this is done on base of the correct DID),
>>> but the problem is, that wrong codices are used if the wrong trunk is
>>> selected.
>>>
>>> Is this a problem of asterisk or is it caused by the provider, which
>>> always addresses the same "trunk" regardless which number has been
>>> called?
>>
>> Asterisk is the one who associates an incoming message with an endpoint.
>> In the case of providers you can use IP based matching - which would
>> behave as you see, only one can be matched. The second option is the
>> line option[1] which may or may not work (it depends on the behavior of
>> the provider). If it works then the right endpoint would be chosen. Out
>> of those two options there's nothing else applicable built in to match.
>>
>> [1]
>> http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
>>
> 
> Unfortunately Deutsche Telekom doesn't support this solution :-(.

Further investigation showed, that Telekom provides the line info in the
Request Line (as seen by Wireshark):

Request-Line: INVITE sip:+49@46.37.15.4:5060;line=azpreyb SIP/2.0

You can't find it if you expect it in contact header - or do you expect
it in the Request-Line?


Regards,
Michael

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Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Michael Maier]

Hello Joshua,

thank you very much for your extremely quick answer! I really appreciate
your work and your friendly and your patient support!


On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
> On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
>> Hello!
>>
>> I've got a problem to select the correct trunk if there is one provider
>> and different numbers with different configurations for this same
>> provider.
>>
>> Example:
>>
>> trunk-prov1-2345
>> trunk-prov1-2346
>> trunk-prov1-2347
>>
>> Each trunk registers an own number (at the same provider) and provides
>> own configuration: they have different allowed codecs e.g..
>>
>> What I'm experiencing now, is, that each incoming call is provided by
>> trunk-prov1-2346, no matter which number has been dialed.
>>
>> The problem isn't the routing (this is done on base of the correct DID),
>> but the problem is, that wrong codices are used if the wrong trunk is
>> selected.
>>
>> Is this a problem of asterisk or is it caused by the provider, which
>> always addresses the same "trunk" regardless which number has been
>> called?
> 
> Asterisk is the one who associates an incoming message with an endpoint.
> In the case of providers you can use IP based matching - which would
> behave as you see, only one can be matched. The second option is the
> line option[1] which may or may not work (it depends on the behavior of
> the provider). If it works then the right endpoint would be chosen. Out
> of those two options there's nothing else applicable built in to match.
> 
> [1]
> http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
> 

Unfortunately Deutsche Telekom doesn't support this solution :-(. I
tried it by faking DNS entries and use different server names to get
different IP addresses. Unfortunately, this doesn't work, because the
name wasn't accepted by the ISP. Using the IP address directly isn't a
good solution, too, because you can't receive changes any more.

A "solution" would be via regularly created rpz DNS entries, which are
used to resolve a fake server name (like fake1.real.name.com) by
asterisk to the desired IP. All other string operations would use the
real.name.com.

Ok, it's a pipe dream!


Regards,
Michael

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Re: [asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Joshua Colp]

On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
> Hello!
> 
> I've got a problem to select the correct trunk if there is one provider
> and different numbers with different configurations for this same
> provider.
> 
> Example:
> 
> trunk-prov1-2345
> trunk-prov1-2346
> trunk-prov1-2347
> 
> Each trunk registers an own number (at the same provider) and provides
> own configuration: they have different allowed codecs e.g..
> 
> What I'm experiencing now, is, that each incoming call is provided by
> trunk-prov1-2346, no matter which number has been dialed.
> 
> The problem isn't the routing (this is done on base of the correct DID),
> but the problem is, that wrong codices are used if the wrong trunk is
> selected.
> 
> Is this a problem of asterisk or is it caused by the provider, which
> always addresses the same "trunk" regardless which number has been
> called?

Asterisk is the one who associates an incoming message with an endpoint.
In the case of providers you can use IP based matching - which would
behave as you see, only one can be matched. The second option is the
line option[1] which may or may not work (it depends on the behavior of
the provider). If it works then the right endpoint would be chosen. Out
of those two options there's nothing else applicable built in to match.

[1]
http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/

-- 
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Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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[asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

[Michael Maier]

Hello!

I've got a problem to select the correct trunk if there is one provider
and different numbers with different configurations for this same provider.

Example:

trunk-prov1-2345
trunk-prov1-2346
trunk-prov1-2347

Each trunk registers an own number (at the same provider) and provides
own configuration: they have different allowed codecs e.g..

What I'm experiencing now, is, that each incoming call is provided by
trunk-prov1-2346, no matter which number has been dialed.

The problem isn't the routing (this is done on base of the correct DID),
but the problem is, that wrong codices are used if the wrong trunk is
selected.

Is this a problem of asterisk or is it caused by the provider, which
always addresses the same "trunk" regardless which number has been called?



Thanks for any hint,
regards,
Michael

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Re: [asterisk-users] log incoming calls without answering

[Frank Vanoni]

On Thu, 2017-04-20 at 17:26 -0300, Fabio Moretti wrote:

> Any idea?

I used to play with an analog telephone line and Asterisk by using a
Linksys SPA-3102 Voice Gateway.

I think it is no longer manufactured, but maybe you con buy a used one
on eBay or you can find an equivalent device from another manufacturer.




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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Thank for all the replies, a lot of input and information!

Sorry for this useless mail, but I really wanted to say thank you.


Il 20/04/2017 17:26, Fabio Moretti ha scritto:
> Hi,
>
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
>
> analog line 1 -+-- asterisk
>|
>\__ analog phone
>
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
>
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
>
> Any idea?
>
> Thanks
>
>
>
>   
>
>

-- 
Fabio Moretti
Gerente de Sistemas
www.tecytal.com 
0800 8780
(+598) 248 77921


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Re: [asterisk-users] log incoming calls without answering

[Tzafrir Cohen]

On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote:
> Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
> >
> > This gets kinda Rube Golberg-ish, but convert the incoming analog line
> > to sip, route it through asterisk and have asterisk do its thing
> > before converting it back to analog to send to the phone. Only problem
> > is you get a lot of extra hardware involved in the mix to make it
> > work. It will be a lot of expense and trouble, so you need to make
> > sure that whatever part you want asterisk to play is worth that
> > effort. Also, I wouldn't touch a fax line in this manner.
> >
> > If you could give a bit more info on what you want asterisk to do, we
> > could maybe give better advice on how to solve your problem.
> 
> Hi Kevin,
> 
> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from "

Simple dialplan. Depending on the type of caller ID system, you may need
to wait a few seconds (in case the caller ID is sent after the first
ring).

Thus, assuming you have a DAHDI device, your dialplan is:

exten => s,1,Wait(5)   ; check how much and if waiting is needed
 same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel 
${CHANNEL(dahdi_channel)})

And move on to report from there.

If you also need to report the total time of the call: that might be
possible if the remote side reverses polarity of the channels on call
start and end. Information about it is currently only reported in debug
messages by chan_dahdi. So it is possible (given polarity reversal), but
tricky.

-- 
   Tzafrir Cohen
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

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Re: [asterisk-users] log incoming calls without answering

[J Montoya or A J Stiles]

On Thursday 20 Apr 2017, Fabio Moretti wrote:
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an
> asterisk for "checking" the inbound calls without answering them, doing
> something like this:
> 
> analog line 1 -+-- asterisk
> 
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date
> and time, etc) somewhere, but nothing more, people will answer using the
> old analog phone. The goal is to have a log of the inbound calls without
> touching the old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok,
> call this AGI, and then don't answer and do nothing more".
> 
> Any idea?

You can get FXO cards, such as the TDM410P/E; which connect to an analogue 
line just like a telephone and then allow the line to be treated as a DAHDI 
channel.  (You would need an FXS card to connect to an analogue telephone; 
this has an additional power supply to generate the 48V DC line voltage and 
80V, 25 Hz ringing voltage.  Usually these cards have 4 or 8 slots for either 
FXO or FXS modules; make sure you buy the right ones.  Explaining to the 
vendor what you want it for may be the best way.)

So you could have Asterisk as an extension to your analogue line, and it could 
monitor incoming calls.  (You would not be able to monitor outgoing calls, 
though; unless you connected each analogue telephone to an FXS port, and used 
Asterisk as a simple PABX.)  

You do not need to Answer() the line to get the caller ID; it comes down the 
line  (either as 300 baud modem tones representing ASCII, or DTMF digit tones, 
depending on your telephone company)  between the polarity reversal and the 
first ring.  And there is no need to have Answer()ed before running an AGI  (in 
fact, you might have good reasons why you would want to have a pre-answer 
script, such as deciding which extension to ring, if you want to direct 
inbound callers straight back to the agent who called them earlier).

All your AGI need do is fork(), and now -- as far as you are concerned -- 
there are two parallel universes, each containing a full copy of the original 
process with the same variables and file handles.  In one, which is the parent, 
fork() returned a non-zero value  (which is the PID of the child process, and 
you might have to clear it out of the process table if it exits before you 
do);  so that process can just exit and return control to Asterisk.  In the 
child process, where the fork() call returned zero  (you don't need to know 
your parent process's PID, since any still-living grandparent will just take 
care of you),  you need just to close STDIN, STDOUT and STDERR  (which are 
still connected to Asterisk)  so it will no longer be waiting for your process 
to do anything -- now you are free!  :)  You then connect to some sort of 
database and run a query like

INSERT INTO call_history(date,source,dest) 
VALUES(NOW(),"${CALLERID(num)}","${EXTEN}"); 

except that ${CALLERID(num)} and ${EXTEN} will have to be passed into the AGI 
script as parameters.  

Then. back in extensions.conf, Dial() a local channel that does nothing but 
Wait() for long enough for the ringing to stop; either because the analogue 
phone has been picked up, or because the caller has stopped trying.  Asterisk 
should get a Hangup() event when the far end hangs up, at any rate.


If you buy a single-channel one of the cheap Chinese clone cards  (from The 
Usual Place),  it ought to last long enough and work well enough for you to 
experiment with; but DO NOT rely on these cheap cards for anything even 
remotely mission-critical.

-- 
JM

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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Re: [asterisk-users] log incoming calls without answering

[Brandon B.]

On 2017-04-20 02:33 PM, Fabio Moretti wrote:

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


Have you considered the legal implications of what you are trying to 
accomplish? You are describing a pen register 
. Asterisk is not suitable 
for this purpose, even if it technically can be used in a fashion. You 
might want look into high impedance telephone line equipment which solve 
the technical issues.


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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

> On 21/04/2017, at 9:33 am, Fabio Moretti  wrote:
> 
> the point basically is: it is possibile for asterisk to log a call
> without answering it? How to do it in the dialplan? Or I'm wasting time
> because an analog line who enter asterisk is always answered?

Yes.

Something similar to this will work (untested, possibly with syntax errors, but 
gives the idea):

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Hangup

The 'Hangup' will simply tell the FXS box (which you've used to terminate the 
line) that Asterisk isn't accepting the call.

One problem could be the FXS box re-inviting Asterisk over and over (causing 
duplicate NoOp logs), but you could add a Wait in there for 2-3 minutes before 
'Hangup'. Not many calls would ring for more than 2-3 minutes. In fact the PSTN 
may limit the number of rings (I have a feeling here in NZ it's about 250 rings 
or so).

exten => s,1,NoOp(Inbound call from callerID $CALLERID(num))
exten => s,n,Wait(180)
exten => s,n,Hangup

Pete



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Re: [asterisk-users] log incoming calls without answering

[Pete Mundy]

Fabio, this doesn't answer your question directly and it's not Asterisk related 
in any way, but it's another way to engineer a solution to the problem and I've 
seen it done before.

Many analog modems will decode the caller ID on the analog line and provide it 
as part of the 'RING' notification on their serial port. This can be used as 
input on a script on a computer to log the caller ID and datestamp.

For the Asterisk solution you propose to be able to work, the analog line has 
to support caller ID. If it does that then it will work with any of these 
modems too.

It's a purely passive connection to the line (just bridge it over the pair) and 
it may be cheaper/simpler/easier for you to build.

Food for thought?

Pete



> On 21/04/2017, at 8:26 am, Fabio Moretti  wrote:
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program an 
> asterisk for "checking" the inbound calls without answering them, doing 
> something like this:
> 
> analog line 1 -+-- asterisk
>   |
>   \__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid, date and 
> time, etc) somewhere, but nothing more, people will answer using the old 
> analog phone.
> The goal is to have a log of the inbound calls without touching the old 
> analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
> this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks


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Re: [asterisk-users] log incoming calls without answering

[James Cloos]

> "FM" == Fabio Moretti  writes:

FM> when a call enter, asterisk sense it and store its values (callerid,
FM> date and time, etc) somewhere, but nothing more, people will answer
FM> using the old analog phone.  The goal is to have a log of the inbound
FM> calls without touching the old analog system (it's shared between
FM> different subjects).

IIUC, the pots line has both some number of analog phones a/o fax
machines on it, plus a fxo->sip gateway, yes?

You can route the sip portion to asterisk and have the dialplan log
everything but never answer.

You may want to call the Ringing dialplan application, but even that
may not be required.  OTOH, calling Ringing should prevent the gateway
from assuming that the asterisk machine never saw the INVITE.

Eventually, when the other extension answers, the fxo->sip gateway will
cancel the sip call just like it would if the caller hung up.

(There is a possibility that any given gateway may not cancel the sip
call until the analog call is completed; you need to test.)

-JimC
-- 
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Re: [asterisk-users] log incoming calls without answering

[John Novack_PIV2]

Fabio Moretti wrote:

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:

I honestly don't know if you can do what you want without some piece
of equipment picking up the line. What I would do is get an analog
line, an analog phone, an analog to sip device (there are many to
choose from) and a basic asterisk instance. I would then make a small
test setup where the analog line goes to a splitter. One side of the
splitter goes to your analog phone. One side goes to your analog to
SIP converter and then into your asterisk instance via your ethernet
network. Use your cell phone to call the number of your analog line
and see if it works. You would have to code a basic dialplan on the
asterisk side and set up the trunk to your converter, which I am
assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?


I used to do something similar, though not for 60 lines. I had ports off a 
channel bank bridged to a propitiatory ( think Vonage ) device, or even a 
copper PSTN line.
Not sure where you are in the world, but different analog lines deliver caller 
id differently, so timing can become an issue, and you will have no clue when 
the call is either answered or abandoned.

Here in the US it is between the first and second ring
You would need to work through the timing, as if the line rings several times 
you need to detect the ring, capture the CLID, then ignore any additional rings 
for x period of time as there will only be one CLID string sent per call.


Once I converted all lines to VOIP it all became much easier.
If your analog lines are NOT in the US or Canada, you may run into different 
issues, or it might even become easier.


John Novack

--
Dog is my co-pilot


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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 18:09, kevin.lar...@pioneerballoon.com ha scritto:
>
> I honestly don't know if you can do what you want without some piece
> of equipment picking up the line. What I would do is get an analog
> line, an analog phone, an analog to sip device (there are many to
> choose from) and a basic asterisk instance. I would then make a small
> test setup where the analog line goes to a splitter. One side of the
> splitter goes to your analog phone. One side goes to your analog to
> SIP converter and then into your asterisk instance via your ethernet
> network. Use your cell phone to call the number of your analog line
> and see if it works. You would have to code a basic dialplan on the
> asterisk side and set up the trunk to your converter, which I am
> assuming you know how to do.

Yes, I'll definitely do the test before set up the whole proyect, but
the point basically is: it is possibile for asterisk to log a call
without answering it? How to do it in the dialplan? Or I'm wasting time
because an analog line who enter asterisk is always answered?







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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from 
"
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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Re: [asterisk-users] log incoming calls without answering

[Fabio Moretti]

Il 20/04/2017 17:32, kevin.lar...@pioneerballoon.com ha scritto:
>
> This gets kinda Rube Golberg-ish, but convert the incoming analog line
> to sip, route it through asterisk and have asterisk do its thing
> before converting it back to analog to send to the phone. Only problem
> is you get a lot of extra hardware involved in the mix to make it
> work. It will be a lot of expense and trouble, so you need to make
> sure that whatever part you want asterisk to play is worth that
> effort. Also, I wouldn't touch a fax line in this manner.
>
> If you could give a bit more info on what you want asterisk to do, we
> could maybe give better advice on how to solve your problem.

Hi Kevin,

I've already proposed your solution (is the most reasonable) but they
have more than 60 analogs lines (no faxes) and some of them terminate in
appliances like alarms, etc, so the solution must not touch in any way
the connection between the line and his termination: doing a analog to
digital conversion, passing it to asterisk and the convert it back to
analog is prone to problems (what if asterisk crashes? or if a gateway
fail?).
I can split the existing lines (there are no complex things like adsl or
digital signaling), convert the branches to digital and terminate then
into an asterisk machine, so any failure will not affect the old
circuit, but of course I've to configure asterisk to ONLY LOG calls and
nothing more.

This is what they want:
- line 1 ring
- line 1 is splitted in two, the first branch (let's say the "analog"
branch) go to an analog phone, that rings
- the second branch go through a gateway and then to asterisk
- asterisk log (with an AGI for example) "line 1 rings at  from "
no more is required from asterisk, if someone answer the analog phone or
not is not my business.







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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> From: Fabio Moretti <fmore...@tecytal.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> <asterisk-users@lists.digium.com>
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program 
> an asterisk for "checking" the inbound calls without answering them,
> doing something like this:
> 
> analog line 1 -+-- asterisk
>|
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid,
> date and time, etc) somewhere, but nothing more, people will answer 
> using the old analog phone.
> The goal is to have a log of the inbound calls without touching the 
> old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: 
> "ok, call this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks

This gets kinda Rube Golberg-ish, but convert the incoming analog line to 
sip, route it through asterisk and have asterisk do its thing before 
converting it back to analog to send to the phone. Only problem is you get 
a lot of extra hardware involved in the mix to make it work. It will be a 
lot of expense and trouble, so you need to make sure that whatever part 
you want asterisk to play is worth that effort. Also, I wouldn't touch a 
fax line in this manner.

If you could give a bit more info on what you want asterisk to do, we 
could maybe give better advice on how to solve your problem.

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[asterisk-users] log incoming calls without answering

[Fabio Moretti]

Hi,

I've some analogic lines and I'm asked if it's possible to program an asterisk 
for "checking" the inbound calls without answering them, doing something like 
this:

analog line 1 -+-- asterisk
   |
   \__ analog phone

when a call enter, asterisk sense it and store its values (callerid, date and 
time, etc) somewhere, but nothing more, people will answer using the old analog 
phone.
The goal is to have a log of the inbound calls without touching the old analog 
system (it's shared between different subjects).

I'm pretty sure it's something possible, but how to tell asterisk: "ok, call 
this AGI, and then don't answer and do nothing more".

Any idea?

Thanks






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[asterisk-users] app_queue missed calls per agent - caller hangup before timeout

[marek cervenka]

hi,

i'm trying get report about missed calls per agent. i'm using queue_log 
and RINGNOANSWER event

but i found problem described here

---
https://www.thirdlane.com/forum/queue-log-problem

RINGNOANSWER only happens if the call TIMES OUT ringing the agent and it 
returns to the queue. If your agent has a 30 second timeout and the 
caller ABANDONS the call in 5 seconds it will log an ABANDON not a 
RINGNOANSWER.


This is the only time ast_queue_log is executed with RINGNOANSWER. The 
subsequent code of this function goes on to autopause the agent/member 
if autopause is enabled. Not something that happens when callers hang up 
when ringing the agents.


/*! \brief RNA == Ring No Answer. Common code that is executed when we 
try a queue member and they don't answer. */
static void rna(int rnatime, struct queue_ent *qe, char *interface, char 
*membername)

{
if (option_verbose > 2)
ast_verbose( VERBOSE_PREFIX_3 "Nobody picked up in %d ms\n", rnatime);
ast_queue_log(qe->parent->name, qe->chan->uniqueid, membername, 
"RINGNOANSWER", "%d", rnatime);

---


any tips howto detect missed calls where caller hangup before timeout?
tnx
Marek

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Re: [asterisk-users] max concurrent calls with bundled pjproject

[George Joseph]

On Thu, Aug 18, 2016 at 3:33 AM, ian gilmour 
wrote:

> Hi,
>
> PJSIP in the past had limitations on the max concurrent calls, etc. There
> were ways to overcome them by changing the source code. (e.g.
> http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.
> org/2013-February/015721.html)
>
> Do any similar tweaks need to be done to the bundled pjproject to handle
> high volumes of concurrent calls with Asterisk?
>
The bundled defaults are already tuned for server type loads.  The
MAX_CALLS and MAX_PLAYERS settings mentioned in that email don't apply to
asterisk since we don't use those components.


> What (if any) are the current default asterisk 13 + pjproject audio +
> video concurrent call limits if using the bundled pjproject + asterisk
> patches as is?
>
Here are the setting we use (third-party/pjproject/patches/config_site.h).
The best optimization is the use of epoll instead of select which we
automatically turn on if the OS is Linux.  It's hard to say what the actual
call limit would be since it's dependent on memory, CPU, etc.  Technically,
it would be 2500 based on MAX_HANDLES but with epoll in use, that number
just controls the size of 1 array and can be easily increased with little
effect on memory utilization.

#include 

#define PJ_HAS_IPV6 1
#define NDEBUG 1
#define PJ_MAX_HOSTNAME (256)
#define PJSIP_MAX_URL_SIZE (512)
#ifdef PJ_HAS_LINUX_EPOLL
#define PJ_IOQUEUE_MAX_HANDLES (5000)
#else
#define PJ_IOQUEUE_MAX_HANDLES (FD_SETSIZE)
#endif
#define PJ_IOQUEUE_HAS_SAFE_UNREG 1
#define PJ_IOQUEUE_MAX_EVENTS_IN_SINGLE_POLL (16)

#define PJ_SCANNER_USE_BITWISE 0
#define PJ_OS_HAS_CHECK_STACK 0
#define PJ_LOG_MAX_LEVEL 3
#define PJ_ENABLE_EXTRA_CHECK 1
#define PJSIP_MAX_TSX_COUNT ((64*1024)-1)
#define PJSIP_MAX_DIALOG_COUNT ((64*1024)-1)
#define PJSIP_UDP_SO_SNDBUF_SIZE (512*1024)
#define PJSIP_UDP_SO_RCVBUF_SIZE (512*1024)
#define PJ_DEBUG 0
#define PJSIP_SAFE_MODULE 0
#define PJ_HAS_STRICMP_ALNUM 0
#define PJ_HASH_USE_OWN_TOLOWER 1
/*
  It is imperative that PJSIP_UNESCAPE_IN_PLACE remain 0 or undefined.
  Enabling it will result in SEGFAULTS when URIs containing escape
sequences are encountered.
*/
#undef PJSIP_UNESCAPE_IN_PLACE
#define PJSIP_MAX_PKT_LEN 6000

#undef PJ_TODO
#define PJ_TODO(x)

/* Defaults too low for WebRTC */
#define PJ_ICE_MAX_CAND 32
#define PJ_ICE_MAX_CHECKS (PJ_ICE_MAX_CAND * 2)


Thanks in advance.
>
> Regards,
>
> IanG
>
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Re: [asterisk-users] max concurrent calls with bundled pjproject

[Marek Červenka]

you can patch it in

[cervenka@matrix asterisk-13.9.1]$ ll third-party/pjproject/
total 24
-rwxrwxr-x. 1 cervenka cervenka  877 May 13 19:41 apply_patches
-rw-rw-r--. 1 cervenka cervenka 1794 May 13 19:41 configure.m4
-rw-rw-r--. 1 cervenka cervenka 5352 May 13 19:41 Makefile
-rw-rw-r--. 1 cervenka cervenka  428 May 13 19:41 Makefile.rules
drwxrwxr-x. 2 cervenka cervenka 4096 May 13 19:41 patches


Dne 18.8.2016 v 11:33 ian gilmour napsal(a):


Hi,

PJSIP in the past had limitations on the max concurrent calls, etc. 
There were ways to overcome them by changing the source code. (e.g. 
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-February/015721.html)


Do any similar tweaks need to be done to the bundled pjproject to 
handle high volumes of concurrent calls with Asterisk?


What (if any) are the current default asterisk 13 + pjproject audio + 
video concurrent call limits if using the bundled pjproject + asterisk 
patches as is?


Thanks in advance.

Regards,

IanG





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[asterisk-users] max concurrent calls with bundled pjproject

[ian gilmour]

Hi,

PJSIP in the past had limitations on the max concurrent calls, etc. There were 
ways to overcome them by changing the source code. (e.g. 
http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2013-February/015721.html
 
)

Do any similar tweaks need to be done to the bundled pjproject to handle high 
volumes of concurrent calls with Asterisk?

What (if any) are the current default asterisk 13 + pjproject audio + video 
concurrent call limits if using the bundled pjproject + asterisk patches as is?

Thanks in advance.

Regards,

IanG-- 
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Re: [asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Looks like it missed 13.9.0 ☹

Thanks,
Derek B.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, May 04, 2016 10:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] Double queue calls being delivered to agents



On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett 
<rmudg...@digium.com<mailto:rmudg...@digium.com>> wrote:


On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
<de...@empire-team.com<mailto:de...@empire-team.com>> wrote:
I posted this over in asterisk-dev, realized I probably should have put it here.

Hi there,
We’ve been having a strange issue with a customer’s queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they’re already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a ‘missed call’.
- Cannot use ‘autopause’ feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried ‘random’ also)
Timeout = 15

We’re on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I’ve scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven’t 
found the actual patch, if one even exists.

I’m hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

This issue has been around a long time and was just recently fixed and I think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115

Looks like it will be in the next release as the issue does not have a target 
release set.
Richard

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Re: [asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Awesome. Thanks again Richard.


On May 4, 2016, at 10:59 PM, Richard Mudgett 
> wrote:



On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett 
> wrote:


On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
> wrote:
I posted this over in asterisk-dev, realized I probably should have put it here.

Hi there,
We've been having a strange issue with a customer's queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they're already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a 'missed call'.
- Cannot use 'autopause' feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried 'random' also)
Timeout = 15

We're on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I've scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven't 
found the actual patch, if one even exists.

I'm hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

This issue has been around a long time and was just recently fixed and I think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115

Looks like it will be in the next release as the issue does not have a target 
release set.

Richard

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Re: [asterisk-users] Double queue calls being delivered to agents

[Richard Mudgett]

On Tue, May 3, 2016 at 8:59 PM, Richard Mudgett  wrote:

>
>
> On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
> wrote:
>
>> I posted this over in asterisk-dev, realized I probably should have put
>> it here.
>>
>> Hi there,
>> We’ve been having a strange issue with a customer’s queues where a queued
>> call will ring an available agent, agent answers, then a second or two
>> later the agent is offered a second call which they cannot answer, since
>> they’re already speaking with a client.
>>
>> This in turn causes a few issues:
>> - Agent stats are no longer accurate, as it gets marked down as a ‘missed
>> call’.
>> - Cannot use ‘autopause’ feature any longer, as the second queue call
>> goes unanswered and pauses the agents.
>>
>> The basic queue setup is as follows:
>> Autofill = yes
>> Ringinuse = no
>> Wrapuptime = 5
>> Strategy = fewestcalls (tried ‘random’ also)
>> Timeout = 15
>>
>> We’re on Asterisk 11.21.2 currently.
>>
>> In talking to a few colleagues, they seem to recall there being an old
>> patch for the Asterisk queues application that inserted a short 100ms delay
>> between delivering first and second calls.  I’ve scoured the web today, and
>> found some old forums posts of people looking for something exactly like
>> this, but haven’t found the actual patch, if one even exists.
>>
>> I’m hoping someone may have some suggestions on some options we can try
>> to eliminate this issue.
>>
>> Thanks for taking the time to read this.
>>
>
> This issue has been around a long time and was just recently fixed and I
> think
> it was just released in the latest v11 version.
> See https://issues.asterisk.org/jira/browse/ASTERISK-16115
>

Looks like it will be in the next release as the issue does not have a
target release set.

Richard
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[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

Sorry for last post -- forgot to wipe out the digest contents :/ 

Derek B

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[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

ll a Sub routine to do a database lookup
> and some other stuff.
> 
> I can't seem to get the syntax right to call Gosub with Originate
> 
> Just testing with the command line I've been unable to make it work with
> any of these attempts:
> 
> originate PJSIP/johntest application Gosub sub-callout s,1
> 
> originate PJSIP/johntest application Gosub sub-callout(s,1)
> 
> originate PJSIP/johntest application Gosub (sub-callout,s,1)
> 
> What Syntax should I be using?
> 
> And if it helps I'll be calling this via AMI over https.
> 
> Thanks!
> 
> -- 
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/6596facc/attachment-0001.html>
> 
> --
> 
> Message: 7
> Date: Tue, 3 May 2016 14:33:32 -0700
> From: Bruce Ferrell <bferr...@baywinds.org>
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Call a subroutine via Originate?
> Message-ID: <9bcf1278-a6c1-b3e5-668d-fd4cdb3f3...@baywinds.org>
> Content-Type: text/plain; charset="windows-1252"; Format="flowed"
> 
> use the macro construct and return from the macro
> 
>> On 5/3/16 2:24 PM, John Kiniston wrote:
>> Howdy everyone,
>> 
>> I'm writing a little click to dial type tool and I've run into a snag 
>> where my Originate command needs to call a Sub routine to do a 
>> database lookup and some other stuff.
>> 
>> I can't seem to get the syntax right to call Gosub with Originate
>> 
>> Just testing with the command line I've been unable to make it work 
>> with any of these attempts:
>> 
>> originate PJSIP/johntest application Gosub sub-callout s,1
>> 
>> originate PJSIP/johntest application Gosub sub-callout(s,1)
>> 
>> originate PJSIP/johntest application Gosub (sub-callout,s,1)
>> 
>> What Syntax should I be using?
>> 
>> And if it helps I'll be calling this via AMI over https.
>> 
>> Thanks!
>> 
>> -- 
>> A human being should be able to change a diaper, plan an invasion, 
>> butcher a hog, conn a ship, design a building, write a sonnet, balance 
>> accounts, build a wall, set a bone, comfort the dying, take orders, 
>> give orders, cooperate, act alone, solve equations, analyze a new 
>> problem, pitch manure, program a computer, cook a tasty meal, fight 
>> efficiently, die gallantly. Specialization is for insects.
>> ---Heinlein
> 
> -- next part --
> An HTML attachment was scrubbed...
> URL: 
> <http://lists.digium.com/pipermail/asterisk-users/attachments/20160503/80a6e460/attachment-0001.html>
> 
> --
> 
> Message: 8
> Date: Tue, 3 May 2016 23:15:53 +
> From: Derek Bolichowski <de...@empire-team.com>
> To: "asterisk-users@lists.digium.com"
><asterisk-users@lists.digium.com>
> Subject: [asterisk-users] Double queue calls being delivered to agents
> Message-ID: <da688326-8edc-470f-9355-f4e5612b0...@empire-team.com>
> Content-Type: text/plain; charset="utf-8"
> 
> I posted this over in asterisk-dev, realized I probably should have put it 
> here. 
> 
> Hi there,
> We?ve been having a strange issue with a customer?s queues where a queued 
> call will ring an available agent, agent answers, then a second or two later 
> the agent is offered a second call which they cannot answer, since they?re 
> already speaking with a client.
> 
> This in turn causes a few issues:
> - Agent stats are no longer accurate, as it gets marked down as a ?missed 
> call?.
> - Cannot use ?autopause? feature any longer, as the second queue call goes 
> unanswered and pauses the agents.
> 
> The basic queue setup is as follows:
> Autofill = yes
> Ringinuse = no
> Wrapuptime = 5
> Strategy = fewestcalls (tried ?random? also)
> Timeout = 15
> 
> We?re on Asterisk 11.21.2 currently.
> 
> In talking to a few colleagues, they seem to recall there being an old patch 
> for the Asterisk queues application that inserted a short 100ms delay between 
> delivering first and second calls.  I?ve scoured the web today, and found 
> some old 

Re: [asterisk-users] Double queue calls being delivered to agents

[Richard Mudgett]

On Tue, May 3, 2016 at 6:15 PM, Derek Bolichowski 
wrote:

> I posted this over in asterisk-dev, realized I probably should have put it
> here.
>
> Hi there,
> We’ve been having a strange issue with a customer’s queues where a queued
> call will ring an available agent, agent answers, then a second or two
> later the agent is offered a second call which they cannot answer, since
> they’re already speaking with a client.
>
> This in turn causes a few issues:
> - Agent stats are no longer accurate, as it gets marked down as a ‘missed
> call’.
> - Cannot use ‘autopause’ feature any longer, as the second queue call goes
> unanswered and pauses the agents.
>
> The basic queue setup is as follows:
> Autofill = yes
> Ringinuse = no
> Wrapuptime = 5
> Strategy = fewestcalls (tried ‘random’ also)
> Timeout = 15
>
> We’re on Asterisk 11.21.2 currently.
>
> In talking to a few colleagues, they seem to recall there being an old
> patch for the Asterisk queues application that inserted a short 100ms delay
> between delivering first and second calls.  I’ve scoured the web today, and
> found some old forums posts of people looking for something exactly like
> this, but haven’t found the actual patch, if one even exists.
>
> I’m hoping someone may have some suggestions on some options we can try to
> eliminate this issue.
>
> Thanks for taking the time to read this.
>

This issue has been around a long time and was just recently fixed and I
think
it was just released in the latest v11 version.
See https://issues.asterisk.org/jira/browse/ASTERISK-16115

Richard
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[asterisk-users] Double queue calls being delivered to agents

[Derek Bolichowski]

I posted this over in asterisk-dev, realized I probably should have put it 
here. 

Hi there,
We’ve been having a strange issue with a customer’s queues where a queued call 
will ring an available agent, agent answers, then a second or two later the 
agent is offered a second call which they cannot answer, since they’re already 
speaking with a client.

This in turn causes a few issues:
- Agent stats are no longer accurate, as it gets marked down as a ‘missed call’.
- Cannot use ‘autopause’ feature any longer, as the second queue call goes 
unanswered and pauses the agents.

The basic queue setup is as follows:
Autofill = yes
Ringinuse = no
Wrapuptime = 5
Strategy = fewestcalls (tried ‘random’ also)
Timeout = 15

We’re on Asterisk 11.21.2 currently.

In talking to a few colleagues, they seem to recall there being an old patch 
for the Asterisk queues application that inserted a short 100ms delay between 
delivering first and second calls.  I’ve scoured the web today, and found some 
old forums posts of people looking for something exactly like this, but haven’t 
found the actual patch, if one even exists.

I’m hoping someone may have some suggestions on some options we can try to 
eliminate this issue.

Thanks for taking the time to read this.

-Derek B
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Re: [asterisk-users] Recording barged calls

[Joshua Colp]

Leandro Dardini wrote:

Hi,
I'd like to record the barged call... but whichever leg of the call I
try to barge, my speaking is never recorded using MixMonitor. Any idea
about the reason?


The only suggestion I have really is to insert a Local channel in the 
mix and record on the real one while barging on the Local.


This is due to a limitation in the underlying mechanism that does not 
allow the position to be defined of where to insert the barging and 
recording. It's fixed so recording comes first, followed by injecting media.


Cheers,

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Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] Recording barged calls

[Leandro Dardini]

Hi,
I'd like to record the barged call... but whichever leg of the call I try
to barge, my speaking is never recorded using MixMonitor. Any idea about
the reason?

Leandro
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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15

[Marlon Araujo]

15 minutes, sure sounds like reinvite could be the villain.

Can you paste your sip.conf

Marlon Araujo

> On Dec 22, 2015, at 13:00, asterisk-users-requ...@lists.digium.com wrote:
> 
> Re: [asterisk-users] Deutsche Telekom: calls dropped after 15

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15

[Luca Bertoncello]

Marlon Araujo  schrieb:

> 15 minutes, sure sounds like reinvite could be the villain.
> 
> Can you paste your sip.conf

Very strange...
I didn't change anything, but now the calls are NOT dropped anymore...
Maybe Telekom changed somewhat...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

Zitat von Sebastian Kemper :


I don't remember seeing anything looking like a SIP trace in your first
mail. Try

sip set debug on

instead of

sip set debug 42

I don't think there's a sip debugging level like 42 in Asterisk. You can
either switch it on or off.


Is it not this:

   http://lists.digium.com/pipermail/asterisk-users/2015-December/288078.html

?

sip set debug 42 should be a little trick to enable more debugging...  
So I got in this list some months ago...


But now somewhat other: yesterday evening I spoke with Telekom. They  
tried to "reset my DSL port" (whatever it means).
As result I was without Internet and phone for over an hour... Then I  
tried to call my cousin in Italy and the call was NOT dropped after 15  
minutes...


I'll try this evening again. Maybe it was a problem by Deutsche Telekom...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Sebastian Kemper]

On Tue, Dec 22, 2015 at 09:30:52AM +, Luca Bertoncello wrote:
> Zitat von Sebastian Kemper :
> 
> Hi Sebastian
> 



> 
> I tried with
> 
> sip set debug 42
> sip set verbose 42
> 
> The result was in my first E-Mail...

Hi Luca,

I don't remember seeing anything looking like a SIP trace in your first
mail. Try

sip set debug on

instead of

sip set debug 42

I don't think there's a sip debugging level like 42 in Asterisk. You can
either switch it on or off.

Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

Zitat von Sebastian Kemper :


No, that's not it. SIP debugging should show you all the SIP messages
like INVITEs, ACKs and the likes. See this link:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Big fat warning: If you want to paste a SIP trace to the mailing list,
make sure to clean it up first (remove passwords, user names, phone
numbers, digest authentication info etc).


OK, I'll try and report to the list

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Sebastian Kemper]

On Tue, Dec 22, 2015 at 09:42:04AM +, Luca Bertoncello wrote:
> Is it not this:
> 
> http://lists.digium.com/pipermail/asterisk-users/2015-December/288078.html
> 
> ?
> 
> sip set debug 42 should be a little trick to enable more debugging...  
> So I got in this list some months ago...

No, that's not it. SIP debugging should show you all the SIP messages
like INVITEs, ACKs and the likes. See this link:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Big fat warning: If you want to paste a SIP trace to the mailing list,
make sure to clean it up first (remove passwords, user names, phone
numbers, digest authentication info etc).

Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Sebastian Kemper]

On Tue, Dec 22, 2015 at 07:19:47AM +0100, Luca Bertoncello wrote:
> "Brian ::"  schrieb:
> 
> > sip trace?
> 
> Could you please explain? I'm not a VoIP-expert...
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)

Hi Luca,

Brian suggests to check the SIP traces. You can either enable SIP
debugging in Asterisk like so:

sip set debug on

Or you could run tcpdump and capture the SIP traffic.

The first option is probably the easiest.

Regards,
Sebastian

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

Zitat von Sebastian Kemper :

Hi Sebastian


Brian suggests to check the SIP traces. You can either enable SIP
debugging in Asterisk like so:

sip set debug on

Or you could run tcpdump and capture the SIP traffic.

The first option is probably the easiest.


I tried with

   sip set debug 42
   sip set verbose 42

The result was in my first E-Mail...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)


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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

Karsten Wemheuer  schrieb:

Hi Karsten!

> the timeout value of 15 minutes directs me to an issue with session
> timer. Try to refuse them by putting the line
> session-timers = refuse
> into the general context of sip.conf. Reload the sip stack with "sip
> reload".

Sorry, I forgot to mention that...
I already have this setting:

session-refresher=uac
session-timers=refuse

> (I assume You are using chan_sip. I don't know how to disable session
> timer in pj sip).

I use chan_sip.

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Brian ::]

sip trace?

On Mon, Dec 21, 2015 at 6:56 PM, Luca Bertoncello 
wrote:

> Karsten Wemheuer  schrieb:
>
> Hi Karsten!
>
> > the timeout value of 15 minutes directs me to an issue with session
> > timer. Try to refuse them by putting the line
> > session-timers = refuse
> > into the general context of sip.conf. Reload the sip stack with "sip
> > reload".
>
> Sorry, I forgot to mention that...
> I already have this setting:
>
> session-refresher=uac
> session-timers=refuse
>
> > (I assume You are using chan_sip. I don't know how to disable session
> > timer in pj sip).
>
> I use chan_sip.
>
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
>
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[asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

Hi list!

My Problem: all calls to international numbers will be dropped after exactly
15 minutes...
I have a VoIP-account by Deutsche Telekom.
This is what I see when I call someone (my parents) and the connection will
be dropped:

  == Using SIP RTP CoS mark 5
-- Executing [+3901522@default:1] Set("SIP/004935-0125", 
"newNumber=003901522") in new stack
-- Executing [+3901522@default:2] 
Verbose("SIP/004935-0125", "2,Rewrite number +3901522 to 
003901522") in new stack
  == Rewrite number +3901522 to 003901522
-- Executing [+3901522@default:3] Dial("SIP/004935-0125", 
"local/003901522") in new stack
-- Called local/003901522
-- Executing [003901522@default:1] 
Verbose("Local/003901522@default-003c;2", "2,DEFAULT") in new stack
  == DEFAULT
-- Executing [003901522@default:2] 
Set("Local/003901522@default-003c;2", "CHANNEL(musicclass)=default") in 
new stack
-- Executing [003901522@default:3] 
GotoIf("Local/003901522@default-003c;2", "0?dialrebvoice") in new stack
-- Executing [003901522@default:4] 
GotoIf("Local/003901522@default-003c;2", "0?dialluca") in new stack
-- Executing [003901522@default:5] 
GotoIf("Local/003901522@default-003c;2", "1?dialluca") in new stack
-- Goto (default,003901522,13)
-- Executing [003901522@default:13] 
Verbose("Local/003901522@default-003c;2", "2,Outgoing call for 
003901522 using pbxluca") in new stack
  == Outgoing call for 003901522 using pbxluca
-- Executing [003901522@default:14] 
Dial("Local/003901522@default-003c;2", 
"SIP/pbxluca/003901522,,RXx") in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/pbxluca/003901522
-- SIP/pbxluca-0126 is ringing
-- SIP/pbxluca-0126 is making progress passing it to 
Local/003901522@default-003c;2
-- Local/003901522@default-003c;1 is ringing
-- Local/003901522@default-003c;1 is making progress passing it to 
SIP/004935-0125
-- SIP/pbxluca-0126 answered Local/003901522@default-003c;2
-- Local/003901522@default-003c;1 answered 
SIP/004935-0125
  == Spawn extension (default, 003901522, 14) exited non-zero on 
'Local/003901522@default-003c;2'
-- fixed jitterbuffer created on channel SIP/004935-0125
  == Spawn extension (default, +3901522, 3) exited non-zero on 
'SIP/004935-0125'
-- fixed jitterbuffer destroyed on channel SIP/004935-0125

My number is the 004935 and I called the 003901522.
Any idea?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Karsten Wemheuer]

Hi Luca,

Am Montag, den 21.12.2015, 18:52 +0100 schrieb Luca Bertoncello: 
> Hi list!
> 
> My Problem: all calls to international numbers will be dropped after exactly
> 15 minutes...
> I have a VoIP-account by Deutsche Telekom.
> This is what I see when I call someone (my parents) and the connection will
> be dropped:
> 
>   == Using SIP RTP CoS mark 5
> -- Executing [+3901522@default:1] Set("SIP/004935-0125", 
> "newNumber=003901522") in new stack
> -- Executing [+3901522@default:2] 
> Verbose("SIP/004935-0125", "2,Rewrite number +3901522 to 
> 003901522") in new stack
>   == Rewrite number +3901522 to 003901522
> -- Executing [+3901522@default:3] Dial("SIP/004935-0125", 
> "local/003901522") in new stack
> -- Called local/003901522
> -- Executing [003901522@default:1] 
> Verbose("Local/003901522@default-003c;2", "2,DEFAULT") in new stack
>   == DEFAULT
> -- Executing [003901522@default:2] 
> Set("Local/003901522@default-003c;2", "CHANNEL(musicclass)=default") 
> in new stack
> -- Executing [003901522@default:3] 
> GotoIf("Local/003901522@default-003c;2", "0?dialrebvoice") in new 
> stack
> -- Executing [003901522@default:4] 
> GotoIf("Local/003901522@default-003c;2", "0?dialluca") in new stack
> -- Executing [003901522@default:5] 
> GotoIf("Local/003901522@default-003c;2", "1?dialluca") in new stack
> -- Goto (default,003901522,13)
> -- Executing [003901522@default:13] 
> Verbose("Local/003901522@default-003c;2", "2,Outgoing call for 
> 003901522 using pbxluca") in new stack
>   == Outgoing call for 003901522 using pbxluca
> -- Executing [003901522@default:14] 
> Dial("Local/003901522@default-003c;2", 
> "SIP/pbxluca/003901522,,RXx") in new stack
>   == Using SIP RTP CoS mark 5
> -- Called SIP/pbxluca/003901522
> -- SIP/pbxluca-0126 is ringing
> -- SIP/pbxluca-0126 is making progress passing it to 
> Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 is ringing
> -- Local/003901522@default-003c;1 is making progress passing it 
> to SIP/004935-0125
> -- SIP/pbxluca-0126 answered Local/003901522@default-003c;2
> -- Local/003901522@default-003c;1 answered 
> SIP/004935-0125
>   == Spawn extension (default, 003901522, 14) exited non-zero on 
> 'Local/003901522@default-003c;2'
> -- fixed jitterbuffer created on channel SIP/004935-0125
>   == Spawn extension (default, +3901522, 3) exited non-zero on 
> 'SIP/004935-0125'
> -- fixed jitterbuffer destroyed on channel SIP/004935-0125
> 
> My number is the 004935 and I called the 003901522.
> Any idea?
> 
> Thanks
> Luca Bertoncello
> (lucab...@lucabert.de)
> 

the timeout value of 15 minutes directs me to an issue with session
timer. Try to refuse them by putting the line
session-timers = refuse
into the general context of sip.conf. Reload the sip stack with "sip
reload".

(I assume You are using chan_sip. I don't know how to disable session
timer in pj sip).

HTH,

Karsten



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Re: [asterisk-users] Deutsche Telekom: calls dropped after 15 minutes

[Luca Bertoncello]

"Brian ::"  schrieb:

> sip trace?

Could you please explain? I'm not a VoIP-expert...

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Recording INCOMING calls

[Rusty Newton]

On Thu, Jul 16, 2015 at 3:37 AM, Luca Bertoncello lucab...@lucabert.de
wrote:

 Hi list!

 I'm trying to configure Asterisk to record incoming calls, if the called
 press *3.
 I added in features.conf:

 automixmon = *3

 then, in my dialplan:

 exten = 1,n,Dial(SIP/004935,20,RcxX)

 Well, if I **CALL** a number I'm able to record the call, but if I'll be
 called, and press *3 nothing happens...


Perhaps the incoming calls are routed through different dialplan and in
that Dial you do not have the proper options? The dialplan you posted
appears to be for dialing an explicit outbound number.


 In the console I can't see anything, too.


You can't see *anything*? You may want to read up on logging configuration
to make sure you have relevant logging channels going to your log file or
console.

https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
https://wiki.asterisk.org/wiki/display/AST/Logging

Once you have verbose output going to a log, make sure it is turned up to 5
and then post the call output to the list. With that we'll be able to see
what is happening.


-- 

Rusty Newton
Digium, Inc. | Community Support Manager445 Jan Davis Drive NW -
Huntsville, AL 35806 - USdirect: +1 256 428 6200
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Re: [asterisk-users] Recording INCOMING calls

[Luca Bertoncello]

Rusty Newton rnew...@digium.com schrieb:

 Perhaps the incoming calls are routed through different dialplan and in
 that Dial you do not have the proper options? The dialplan you posted
 appears to be for dialing an explicit outbound number.

YES!! That was the problem!

I just added xX to the previous Dial and all work!

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Recording INCOMING calls

[Rusty Newton]

On Fri, Jul 17, 2015 at 9:09 AM, Luca Bertoncello lucab...@lucabert.de
wrote:

 Rusty Newton rnew...@digium.com schrieb:

  Perhaps the incoming calls are routed through different dialplan and in
  that Dial you do not have the proper options? The dialplan you posted
  appears to be for dialing an explicit outbound number.

 YES!! That was the problem!

 I just added xX to the previous Dial and all work!


Awesome. Be aware that using both x and X will allow both parties on the
call to start recording.. that means regardless of whether this is an
inbound or outbound call that both the calling and the called party will be
able to initiate recording (assuming they know the code, or press it
accidentally).

-- 

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Huntsville, AL 35806 - USdirect: +1 256 428 6200
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[asterisk-users] Recording INCOMING calls

[Luca Bertoncello]

Hi list!

I'm trying to configure Asterisk to record incoming calls, if the called
press *3.
I added in features.conf:

automixmon = *3

then, in my dialplan:

exten = 1,n,Dial(SIP/004935,20,RcxX)

Well, if I **CALL** a number I'm able to record the call, but if I'll be
called, and press *3 nothing happens...
In the console I can't see anything, too.

Could you suggest me what is wrong?

Thanks
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Salaheddine Elharit]

what about

exten = s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)

regards

2015-04-08 5:45 GMT+00:00 Dmitriy Serov serov@gmail.com:

  Hi, Andrew.

 You are trying to solve two tasks: definition through what line the call
 came and a beautiful display of this information.
 1. definition through what line the call came. If the username and
 password for inbound and outbound registration the same, then try the
 following:
 a) delete register lines.
 b) add option callbackextension=Company1 to Company1 friend section..
 And in others with their names too.
 or you can change /s to /Company1 in register line.

 2. beautiful display of this information
 a) add option setvar=fromCompany=Company1 to Company1 friend section..
 b) In dialplan add
 Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})

 Maybe this will help?

 Dmitiy.

 08.04.2015 2:48, Andrew Galdes пишет:

 Hi Dmitriy and others and thanks for your help so far.

  The option match_auth_username=yes seems to have had no effect. From
 my reading, this option will try to match the username of the incoming SIP
 account to a section heading. If that is how it must work then i can see a
 big problem. I'm trying to present the receptionist with a nice display of
 which line the call came in on. For example, the receptionist answers calls
 for 8 different companies and would like the phone to display the company
 name that she should announce to the caller.

  Here is a more complete output of an incoming call. I've changed the SIP
 numbers to Company1', etc, to hide the numbers.

  Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
 Verbosity is at least 12
 asterisk*CLI
 asterisk*CLI
 asterisk*CLI
   == Using SIP RTP CoS mark 5
 -- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*, 
 *thedid=NodePhonesip:compa...@sip.internode.on.net
 sip%3acompa...@sip.internode.on.net*) in new stack
 -- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
 http://sip.internode.on.net*) in new stack
 -- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2*) in new stack
 -- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=** sip:Company2*) in new stack
 -- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,33,1:6*) in new stack
 -- Goto (incoming,s,6)
 -- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,88,1:7*) in new stack
 -- Goto (incoming,s,7)
 -- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,36,1:8*) in new stack
 -- Goto (incoming,s,8)
 -- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*, 
 *1?internal,36,1:9*) in new stack
 -- Goto (internal,36,1)
 -- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*, 
 *CALLERID(name)=SIP/**Company1**-0797*) in new stack
 -- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*, 
 *SIP/36,20*) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/36
 -- SIP/36-0798 is ringing
   == Spawn extension (internal, 36, 2) exited non-zero on
 'SIP/Company1-0797'
 asterisk*CLI exit


  And here is the sip.conf:

  [general]
 match_auth_username=yes
 register=081...:...@sip.internode.on.net/s
 register=082...:...@sip.internode.on.net/s
 register=083...:...@sip.internode.on.net:/s
 register=084...:...@sip.internode.on.net:/s
 register=085...:...@sip.internode.on.net/s
 register=086...:...@sip.internode.on.net/s
 register=087...:...@sip.internode.on.net/s
 register=088...:...@sip.internode.on.net/s

 [Company1]
 username=081...
 fromuser=081...
 secret=...
 canreinvite=no
 qualify=yes
 context=incoming
 type=friend
 insecure=invite,port
 fromdomain=sip.internode.on.net
 host=sip.internode.on.net
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 allow=ulaw
 allow=g729
 bindport=5060
 bindaddr=0.0.0.0
 nat=yes
 registertimeout=5
 allowoverlap=no
 srvlookup=no
 ubscribecontext=from-sip
 callcounter=yes



 [Company2]
 ...
 [Company3]
 ...
 [Company4]
 ...

   And here is some of the extensions.conf file:

  [incoming]
 ; Get the DID number from the TO header.
 exten = s,1,Set(thedid=${SIP_HEADER(TO)})
 exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
 exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
 exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})


 ; Direct the DID accordingly.
 exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
 exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
 exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
 exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
 exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
 exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
 exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
 exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)



  -Andrew 

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[John Kiniston]

Andrew,

Instead of your SET and GOTO blocks I'd recommend using the Asterisk DB to
make things easier to maintain.

You could make two database entries for each of your DID's

database put 4259981810 name JohnPersonal
database put 4259981810 target kiniston-extern,john-personal,1

Then you could do a single block that would do the lookup and call routing:
Set(DESTINATION=${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)})
Set(CALLERID(name)=${DB(${DESTINATION}/name)})
Goto(${DB(${DESTINATION}/target)})


On Tue, Apr 7, 2015 at 6:06 PM, Andrew Galdes andrew.gal...@agix.com.au
wrote:

 Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
 it does work. For prosperity, the SIP service is through Internode.

 Here is my extensions.conf file:

 exten = s,5,Set(callersname=${IF($[ ${pseudodid} =
 081...]?Company1:${callersname})})
 exten = s,6,Set(callersname=${IF($[ ${pseudodid}
 = 082...]?Company2:${callersname})})

 exten = s,13,GotoIf($[${callersname} = Company1]?internal,36,1:14);
 to reception
 exten = s,14,GotoIf($[${callersname} = Company2]?internal,88,1:15);
 to department1

 And later in same file:

 ; Phone 36 reception
 *exten = 36,1,Set(CALLERID(name)=${callersname})*
 exten = 36,n,Dial(SIP/36,20)
 exten = 36,n,VoiceMail(36,u)
 exten = 36,n,Hangup



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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Hi Dmitriy and others and thanks for your help so far.

The option match_auth_username=yes seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on. For example, the receptionist answers calls
for 8 different companies and would like the phone to display the company
name that she should announce to the caller.

Here is a more complete output of an incoming call. I've changed the SIP
numbers to Company1', etc, to hide the numbers.

Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
 Verbosity is at least 12
 asterisk*CLI
 asterisk*CLI
 asterisk*CLI
   == Using SIP RTP CoS mark 5
 -- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*, 
 *thedid=NodePhonesip:compa...@sip.internode.on.net
 sip%3acompa...@sip.internode.on.net*) in new stack
 -- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
 http://sip.internode.on.net*) in new stack
 -- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2*) in new stack
 -- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=** sip:Company2*) in new stack
 -- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,33,1:6*) in new stack
 -- Goto (incoming,s,6)
 -- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,88,1:7*) in new stack
 -- Goto (incoming,s,7)
 -- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,36,1:8*) in new stack
 -- Goto (incoming,s,8)
 -- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*, 
 *1?internal,36,1:9*) in new stack
 -- Goto (internal,36,1)
 -- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*, 
 *CALLERID(name)=SIP/**Company1**-0797*) in new stack
 -- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*, 
 *SIP/36,20*) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/36
 -- SIP/36-0798 is ringing
   == Spawn extension (internal, 36, 2) exited non-zero on
 'SIP/Company1-0797'
 asterisk*CLI exit


And here is the sip.conf:

[general]
 match_auth_username=yes
 register=081...:...@sip.internode.on.net/s
 register=082...:...@sip.internode.on.net/s
 register=083...:...@sip.internode.on.net:/s
 register=084...:...@sip.internode.on.net:/s
 register=085...:...@sip.internode.on.net/s
 register=086...:...@sip.internode.on.net/s
 register=087...:...@sip.internode.on.net/s
 register=088...:...@sip.internode.on.net/s

 [Company1]
 username=081...
 fromuser=081...
 secret=...
 canreinvite=no
 qualify=yes
 context=incoming
 type=friend
 insecure=invite,port
 fromdomain=sip.internode.on.net
 host=sip.internode.on.net
 dtmfmode=rfc2833
 disallow=all
 allow=alaw
 allow=ulaw
 allow=g729
 bindport=5060
 bindaddr=0.0.0.0
 nat=yes
 registertimeout=5
 allowoverlap=no
 srvlookup=no
 ubscribecontext=from-sip
 callcounter=yes



[Company2]
 ...
 [Company3]
 ...
 [Company4]
 ...

 And here is some of the extensions.conf file:

[incoming]
 ; Get the DID number from the TO header.
 exten = s,1,Set(thedid=${SIP_HEADER(TO)})
 exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
 exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
 exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})


 ; Direct the DID accordingly.
 exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
 exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
 exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
 exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
 exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
 exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
 exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
 exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)



-Andrew Galdes


On Thu, Apr 2, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com wrote:


 This is one of the chronic problems. Try this option in sip.conf:
 match_auth_username=yes

 Carefully read the description, it is better to test in after hours.

 02.04.2015 2:50, Andrew Galdes пишет:

 Hello all,

  I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
 with the same service provides. We have 8 phone numbers in total.

  Incoming calls from the public are all correctly directed to appropriate
 office handsets. However, the display on the reception phone (the only one
 i care about) is always showing the same SIP/Account1_0843214321 rather
 than the account representing the number dialed.

  For-instance, if Sam on her mobile calls *08*, Asterisk will
 show a log entry like the following:

  -- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
 

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.

Here is my extensions.conf file:

exten = s,1,Set(thedid=${SIP_HEADER(TO)}); ignore this one
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})

exten = s,5,Set(callersname=${IF($[ ${pseudodid} =
081...]?Company1:${callersname})})
exten = s,6,Set(callersname=${IF($[ ${pseudodid}
= 082...]?Company2:${callersname})})
exten = s,7,Set(callersname=${IF($[ ${pseudodid}
= 083...]?Company3:${callersname})})
exten = s,8,Set(callersname=${IF($[ ${pseudodid}
= 084...]?Company4:${callersname})})
exten = s,9,Set(callersname=${IF($[ ${pseudodid}
= 085...]?Company5:${callersname})})
exten = s,10,Set(callersname=${IF($[ ${pseudodid}
= 086...]?Company6:${callersname})})
exten = s,11,Set(callersname=${IF($[ ${pseudodid}
= 087...]?Company7:${callersname})})
exten = s,12,Set(callersname=${IF($[ ${pseudodid}
= 088...]?Company8:${callersname})})

exten = s,13,GotoIf($[${callersname} = Company1]?internal,36,1:14); to
reception
exten = s,14,GotoIf($[${callersname} = Company2]?internal,88,1:15); to
department1
exten = s,15,GotoIf($[${callersname} = Company3]?internal,36,1:16); to
reception
exten = s,16,GotoIf($[${callersname} = Company4]?internal,36,1:17); to
reception
exten = s,17,GotoIf($[${callersname} = Company5]?internal,36,1:18); to
reception
exten = s,18,GotoIf($[${callersname} = Company6]?internal,89,1:19); to
department2
exten = s,19,GotoIf($[${callersname} = Company7]?internal,36,1:20); to
reception
exten = s,20,GotoIf($[${callersname} = Company8]?internal,13,1:21); to
department3

And later in same file:

; Phone 36 reception
 *exten = 36,1,Set(CALLERID(name)=${callersname})*
 exten = 36,n,Dial(SIP/36,20)
 exten = 36,n,VoiceMail(36,u)
 exten = 36,n,Hangup


Ta,


-Andrew Galdes
Managing Director

RHCE, LPI, CCENT

AGIX Linux

Ph: 08 7324 4429
Mb: 0422 927 598

Find us: Website http://www.agix.com.au | LinkedIn
http://au.linkedin.com/in/andrewgaldes | Blog http://agix.com.au/blog |
YouTube http://www.youtube.com/user/andrewgaldes | Google+
http://google.com/+AndrewGaldes

*Platform Architects for High Demand Web Applications.*

On Wed, Apr 8, 2015 at 9:18 AM, Andrew Galdes andrew.gal...@agix.com.au
wrote:

 Hi Dmitriy and others and thanks for your help so far.

 The option match_auth_username=yes seems to have had no effect. From my
 reading, this option will try to match the username of the incoming SIP
 account to a section heading. If that is how it must work then i can see a
 big problem. I'm trying to present the receptionist with a nice display of
 which line the call came in on. For example, the receptionist answers calls
 for 8 different companies and would like the phone to display the company
 name that she should announce to the caller.

 Here is a more complete output of an incoming call. I've changed the SIP
 numbers to Company1', etc, to hide the numbers.

 Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267)
 Verbosity is at least 12
 asterisk*CLI
 asterisk*CLI
 asterisk*CLI
   == Using SIP RTP CoS mark 5
 -- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*, 
 *thedid=NodePhonesip:compa...@sip.internode.on.net
 sip%3acompa...@sip.internode.on.net*) in new stack
 -- Executing [s@incoming:2] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
 http://sip.internode.on.net*) in new stack
 -- Executing [s@incoming:3] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=NodePhonesip:** sip:Company2*) in new stack
 -- Executing [s@incoming:4] *Set*(*SIP/**Company1**-0797*, 
 *pseudodid=** sip:Company2*) in new stack
 -- Executing [s@incoming:5] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,33,1:6*) in new stack
 -- Goto (incoming,s,6)
 -- Executing [s@incoming:6] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,88,1:7*) in new stack
 -- Goto (incoming,s,7)
 -- Executing [s@incoming:7] *GotoIf*(*SIP/**Company1**-0797*, 
 *0?internal,36,1:8*) in new stack
 -- Goto (incoming,s,8)
 -- Executing [s@incoming:8] *GotoIf*(*SIP/**Company1**-0797*, 
 *1?internal,36,1:9*) in new stack
 -- Goto (internal,36,1)
 -- Executing [36@internal:1] *Set*(*SIP/**Company1**-0797*, 
 *CALLERID(name)=SIP/**Company1**-0797*) in new stack
 -- Executing [36@internal:2] *Dial*(*SIP/**Company1**-0797*, 
 *SIP/36,20*) in new stack
   == Using SIP RTP CoS mark 5
 -- Called SIP/36
 -- SIP/36-0798 is ringing
   == Spawn extension (internal, 36, 2) exited non-zero on
 'SIP/Company1-0797'
 asterisk*CLI exit


 And here is the sip.conf:

 [general]
 match_auth_username=yes
 register=081...:...@sip.internode.on.net/s
 register=082...:...@sip.internode.on.net/s
 register=083...:...@sip.internode.on.net:/s
 

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Dmitriy Serov]

Hi, Andrew.

You are trying to solve two tasks: definition through what line the call 
came and a beautiful display of this information.
1. definition through what line the call came. If the username and 
password for inbound and outbound registration the same, then try the 
following:

a) delete register lines.
b) add option callbackextension=Company1 to Company1 friend section.. 
And in others with their names too.

or you can change /s to /Company1 in register line.

2. beautiful display of this information
a) add option setvar=fromCompany=Company1 to Company1 friend section..
b) In dialplan add
Set(CALLERID(name)=${fromCompany} ${CALLERID(name)})

Maybe this will help?

Dmitiy.

08.04.2015 2:48, Andrew Galdes пишет:

Hi Dmitriy and others and thanks for your help so far.

The option match_auth_username=yes seems to have had no effect. From 
my reading, this option will try to match the username of the incoming 
SIP account to a section heading. If that is how it must work then i 
can see a big problem. I'm trying to present the receptionist with a 
nice display of which line the call came in on. For example, the 
receptionist answers calls for 8 different companies and would like 
the phone to display the company name that she should announce to the 
caller.


Here is a more complete output of an incoming call. I've changed the 
SIP numbers to Company1', etc, to hide the numbers.


Connected to Asterisk 10.12.4 currently running on asterisk (pid =
32267)
Verbosity is at least 12
asterisk*CLI
asterisk*CLI
asterisk*CLI
  == Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*,
*thedid=NodePhonesip:compa...@sip.internode.on.net
mailto:sip%3acompa...@sip.internode.on.net*) in new stack
-- Executing [s@incoming:2]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
http://sip.internode.on.net*) in new stack
-- Executing [s@incoming:3]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2*) in new stack
-- Executing [s@incoming:4]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=** sip:Company2*) in new stack
-- Executing [s@incoming:5]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,33,1:6*) in
new stack
-- Goto (incoming,s,6)
-- Executing [s@incoming:6]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,88,1:7*) in
new stack
-- Goto (incoming,s,7)
-- Executing [s@incoming:7]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,36,1:8*) in
new stack
-- Goto (incoming,s,8)
-- Executing [s@incoming:8]
*GotoIf*(*SIP/**Company1**-0797*, *1?internal,36,1:9*) in
new stack
-- Goto (internal,36,1)
-- Executing [36@internal:1]
*Set*(*SIP/**Company1**-0797*,
*CALLERID(name)=SIP/**Company1**-0797*) in new stack
-- Executing [36@internal:2]
*Dial*(*SIP/**Company1**-0797*, *SIP/36,20*) in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/36
-- SIP/36-0798 is ringing
  == Spawn extension (internal, 36, 2) exited non-zero on
'SIP/Company1-0797'
asterisk*CLI exit


And here is the sip.conf:

[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
http://081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
http://082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
http://085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
http://086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
http://087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
http://088...:...@sip.internode.on.net/s

[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net http://sip.internode.on.net
host=sip.internode.on.net http://sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes

[Company2]
...
[Company3]
...
[Company4]
...

And here is some of the extensions.conf file:

[incoming]
; Get the DID number from the TO header.
exten = s,1,Set(thedid=${SIP_HEADER(TO)})
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})


; Direct the DID accordingly.
exten = s,5,GotoIf($[${pseudodid} = 

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andres]

On 4/7/15 7:48 PM, Andrew Galdes wrote:

Hi Dmitriy and others and thanks for your help so far.

The option match_auth_username=yes seems to have had no effect. From 
my reading, this option will try to match the username of the incoming 
SIP account to a section heading. If that is how it must work then i 
can see a big problem. I'm trying to present the receptionist with a 
nice display of which line the call came in on. For example, the 
receptionist answers calls for 8 different companies and would like 
the phone to display the company name that she should announce to the 
caller.


Here is a more complete output of an incoming call. I've changed the 
SIP numbers to Company1', etc, to hide the numbers.


Connected to Asterisk 10.12.4 currently running on asterisk (pid =
32267)
Verbosity is at least 12
asterisk*CLI
asterisk*CLI
asterisk*CLI
  == Using SIP RTP CoS mark 5
-- Executing [s@incoming:1] *Set*(*SIP/Company1-0797*,
*thedid=NodePhonesip:compa...@sip.internode.on.net
mailto:sip%3acompa...@sip.internode.on.net*) in new stack
-- Executing [s@incoming:2]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2**@sip.internode.on.net
http://sip.internode.on.net*) in new stack
-- Executing [s@incoming:3]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=NodePhonesip:** sip:Company2*) in new stack
-- Executing [s@incoming:4]
*Set*(*SIP/**Company1**-0797*,
*pseudodid=** sip:Company2*) in new stack
-- Executing [s@incoming:5]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,33,1:6*) in
new stack
-- Goto (incoming,s,6)
-- Executing [s@incoming:6]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,88,1:7*) in
new stack
-- Goto (incoming,s,7)
-- Executing [s@incoming:7]
*GotoIf*(*SIP/**Company1**-0797*, *0?internal,36,1:8*) in
new stack
-- Goto (incoming,s,8)
-- Executing [s@incoming:8]
*GotoIf*(*SIP/**Company1**-0797*, *1?internal,36,1:9*) in
new stack
-- Goto (internal,36,1)
-- Executing [36@internal:1]
*Set*(*SIP/**Company1**-0797*,
*CALLERID(name)=SIP/**Company1**-0797*) in new stack
-- Executing [36@internal:2]
*Dial*(*SIP/**Company1**-0797*, *SIP/36,20*) in new stack
  == Using SIP RTP CoS mark 5
-- Called SIP/36
-- SIP/36-0798 is ringing
  == Spawn extension (internal, 36, 2) exited non-zero on
'SIP/Company1-0797'
asterisk*CLI exit


And here is the sip.conf:

[general]
match_auth_username=yes
register=081...:...@sip.internode.on.net/s
http://081...:...@sip.internode.on.net/s
register=082...:...@sip.internode.on.net/s
http://082...:...@sip.internode.on.net/s
register=083...:...@sip.internode.on.net:/s
register=084...:...@sip.internode.on.net:/s
register=085...:...@sip.internode.on.net/s
http://085...:...@sip.internode.on.net/s
register=086...:...@sip.internode.on.net/s
http://086...:...@sip.internode.on.net/s
register=087...:...@sip.internode.on.net/s
http://087...:...@sip.internode.on.net/s
register=088...:...@sip.internode.on.net/s
http://088...:...@sip.internode.on.net/s

[Company1]
username=081...
fromuser=081...
secret=...
canreinvite=no
qualify=yes
context=incoming
type=friend
insecure=invite,port
fromdomain=sip.internode.on.net http://sip.internode.on.net
host=sip.internode.on.net http://sip.internode.on.net
dtmfmode=rfc2833
disallow=all
allow=alaw
allow=ulaw
allow=g729
bindport=5060
bindaddr=0.0.0.0
nat=yes
registertimeout=5
allowoverlap=no
srvlookup=no
ubscribecontext=from-sip
callcounter=yes

[Company2]
...
[Company3]
...
[Company4]
...

And here is some of the extensions.conf file:

[incoming]
; Get the DID number from the TO header.
exten = s,1,Set(thedid=${SIP_HEADER(TO)})
exten = s,2,Set(pseudodid=${SIP_HEADER(To)})
exten = s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten = s,4,Set(pseudodid=${CUT(pseudodid,:,2)})


; Direct the DID accordingly.
exten = s,5,GotoIf($[${pseudodid} = 081]?internal,33,1:6)
exten = s,6,GotoIf($[${pseudodid} = 082]?internal,88,1:7)
exten = s,7,GotoIf($[${pseudodid} = 083]?internal,36,1:8)
exten = s,8,GotoIf($[${pseudodid} = 084]?internal,36,1:9)
exten = s,9,GotoIf($[${pseudodid} = 085]?internal,36,1:10)
exten = s,10,GotoIf($[${pseudodid} = 086]?internal,89,1:11)
exten = s,11,GotoIf($[${pseudodid} = 087]?internal,36,1:12)
exten = s,12,GotoIf($[${pseudodid} = 088]?internal,13,1:13)


Since your objective is to have the receptionist identify the company 
she should be answering to then might I suggest a simple workaround to 
your problem.  Since right here you are already sending the call to the 
expected internal context and 

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andres]

On 4/1/15 7:50 PM, Andrew Galdes wrote:

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip 
accounts with the same service provides. We have 8 phone numbers in 
total.


Incoming calls from the public are all correctly directed to 
appropriate office handsets. However, the display on the reception 
phone (the only one i care about) is always showing the same 
SIP/Account1_0843214321 rather than the account representing the 
number dialed.


For-instance, if Sam on her mobile calls *08*, Asterisk will 
show a log entry like the following:


-- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
thedid=NodePhonesip:*08*@sip.internode.on.net 
http://sip.internode.on.net) in new stack


But Account1_*08* (as the name suggests) has a phone number 
of *08* and not *08*.


It looks like all incoming calls are all being matched against the same 
entry in sip.conf.   A 'set set debug on' should clearly indicate this.  
Look for the line that says :  Found peer 'insert peer name here' for 
'08'
So Sam's call will come through and be routed to the correct handset 
as the business needs, but it seems that all incoming calls are being 
labeled as though coming in on a different account. The effective 
problem is that the calledID is now wrong.


I'm after some general advice on how to handle the problem.

Ta,


-Andrew





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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Dmitriy Serov]

This is one of the chronic problems. Try this option in sip.conf:
match_auth_username=yes

Carefully read the description, it is better to test in after hours.

02.04.2015 2:50, Andrew Galdes пишет:

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip 
accounts with the same service provides. We have 8 phone numbers in 
total.


Incoming calls from the public are all correctly directed to 
appropriate office handsets. However, the display on the reception 
phone (the only one i care about) is always showing the same 
SIP/Account1_0843214321 rather than the account representing the 
number dialed.


For-instance, if Sam on her mobile calls *08*, Asterisk will 
show a log entry like the following:


-- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
thedid=NodePhonesip:*08*@sip.internode.on.net 
http://sip.internode.on.net) in new stack


But Account1_*08* (as the name suggests) has a phone number 
of *08* and not *08*.


So Sam's call will come through and be routed to the correct handset 
as the business needs, but it seems that all incoming calls are being 
labeled as though coming in on a different account. The effective 
problem is that the calledID is now wrong.


I'm after some general advice on how to handle the problem.

Ta,


-Andrew




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[asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[Andrew Galdes]

Hello all,

I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
with the same service provides. We have 8 phone numbers in total.

Incoming calls from the public are all correctly directed to appropriate
office handsets. However, the display on the reception phone (the only one
i care about) is always showing the same SIP/Account1_0843214321 rather
than the account representing the number dialed.

For-instance, if Sam on her mobile calls *08*, Asterisk will show
a log entry like the following:

-- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
thedid=NodePhonesip:*08*@sip.internode.on.net) in new stack
But Account1_*08* (as the name suggests) has a phone number of 
*08* and not *08*.

So Sam's call will come through and be routed to the correct handset as the
business needs, but it seems that all incoming calls are being labeled as
though coming in on a different account. The effective problem is that the
calledID is now wrong.

I'm after some general advice on how to handle the problem.

Ta,


-Andrew
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Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

[John Kiniston]

Can you show us the CDR record for that call?

And maybe what your s priority of your incoming context is?

It should be easy to get what number was dialed, Try:

${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)}

Normally I display the callers number on my phones, Not the number they
dialed?

On Wed, Apr 1, 2015 at 4:50 PM, Andrew Galdes andrew.gal...@agix.com.au
wrote:

 Hello all,

 I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts
 with the same service provides. We have 8 phone numbers in total.

 Incoming calls from the public are all correctly directed to appropriate
 office handsets. However, the display on the reception phone (the only one
 i care about) is always showing the same SIP/Account1_0843214321 rather
 than the account representing the number dialed.

 For-instance, if Sam on her mobile calls *08*, Asterisk will
 show a log entry like the following:

 -- Executing [s@incoming:1] Set(SIP/*Account1_08*, 
 thedid=NodePhonesip:*08*@sip.internode.on.net) in new stack
 But Account1_*08* (as the name suggests) has a phone number of 
 *08* and not *08*.

 So Sam's call will come through and be routed to the correct handset as
 the business needs, but it seems that all incoming calls are being labeled
 as though coming in on a different account. The effective problem is that
 the calledID is now wrong.

 I'm after some general advice on how to handle the problem.

 Ta,


 -Andrew

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A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
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Re: [asterisk-users] Anonymous SIP calls

[James Cloos]

Some of us do allow sip from the internet, but just like for smtp email
protections are in order.

I point my SRV records at dedicated sip proxies (I use kamailio) which
check the INVITEd sip uri the same way my MXs check the SMTP Evelope-To
addresses, and only allow INVITEs through to authorized destinations.

And when those INVITEs make it to asterisk/freeswitch or the like, the
dialplan is generally not direct to phone(s), but via an IVR.

As an example, calling my email address via sip goes to an Asterisk
FollowMe instance.

I also provide my clients with dedicated sip addresses which avoid the
protections.

But the vast majority of the INVITEs coming to my public sip proxies are
fraud attempts.  My primary sip proxy has blocked over 32k fraudulent
INVITEs over the last six months.  And about one OPTIONS sip:100@... per
hour by something calling itself friendly-scanner.

Then again, the number of invalid sip INVITEs per public sip destination
are fewer than the number of spam/virus type SMTP attempts per unit time.

And all of the telemarking fraud I have had to deal with have come via
pstn dids, not via direct sip.

A half-gig virtual works fine for such a sip proxy.

You may also want to look into getting an ISN number, check out
http://freenum.org/ for the details.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] Anonymous SIP calls

[James B. Byrne]

On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
 You have to consider whether you really want anonymous calls, or you
 just want to enable SIP calls from trusted companies/partners.  The
 latter means setting up routes to these companies and (ideally)
 registration between peers.


This is what I am trying to get a handle on.  It seemed to me that the
promise of VOIP was essentially that one could use the Internet as a
replacement for the PSTN directly, providing that ones callers/callees
were also directly connected via VOIP.  SIP providers I had considered
a necessary transition to act as gateways between PSTN dialing and
VOIP until VOIP replaced PSTN virtually entirely if not completely.

That is why we are on Asterisk.  We had to replace our old keyed
system and the thought was that we might as well get ready for VOIP
even if we planned to stay on PSTN for the foreseeable future.

However, the overwhelming evidence I find is that one simply does not
employ VOIP in the same way that PSTN works.  Actually, I have put
that backwards.  What I have discovered is that the most commonly
recommended method is to switch from a Telco to A SIP provider and
continue in a manner similar to the former set-up.  External calls all
have to travel through a third party provider.

One does not accept incoming VOIP calls from just everyone,
apparently.  One only accepts VOIP calls from known correspondents.  I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.

Even limiting VOIP to known correspondents one is ultimately trusting
that they themselves are secured sufficiently to prevent unauthorised
access to your systems through theirs.  And that seems a bit of a
stretch by way of rationalisation to me.

Also I do not understand is why the same issues do not exist from
incoming calls via PSTN.

I somewhat understand the process of getting devices to register and
authenticate to obtain access to our outgoing routes.   What is it
about incoming SIP calls destined to our internal users that make
those calls so dangerous?  Why cannot incoming anonymous SIP calls not
be treated exactly as incoming PSTN calls (other than PSTN have to go
though DAHDI to turn them into digital VOIP calls). What is it that
prevents them from being blocked from gatewaying through to our PSTN
lines?

Please forgive my abysmal ignorance on this matter.  Perhaps I have
been down in the weeds too long getting our internal FreePBX system
working to see what is obvious to others.  I have been going theough
the Asticon Videos on security and have or already had implemented
most of the suggestions: Outbound LD secured by pins and allowed only
during work hours; IPTABLES rules and fail2ban checks; Separation of
voice and data network segments and addresses; Private IP for VOIP
desk-sets and internal provisioning; and so forth.

However, I still have the sense that I am just not getting it.  What
am I missing?

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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