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[asterisk-users] Signaling incoming call

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lucabert at lucabert.de
Guest

Posted: Sun May 31, 2015 1:31 am Post subject: [asterisk-users] Signaling incoming call

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone will receive calls from 3 numbers. All that was done in my dialplan. Now, it would be nice, if I can signaling on the phone which number will be called, so that, for example, if I receive a call for +493511111111 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +493512222222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. Thanks a lot Luca Bertoncello (lucabert@lucabert.de) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

webaccounts173 at jgoe...
Guest

Posted: Sun May 31, 2015 3:26 am Post subject: [asterisk-users] Signaling incoming call

Quote:

Finally I got my Asterisk works with my two phones...
It was a problem on my Firewall (for the phone of my wife) and on my Dialplan
(for forwarding calls).

Now all works as expected, at least in the simulation I did with AsteriskNOW.
Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP...

Well, now I have some time to spend with "fooling"...
My phone will receive calls from 3 numbers. All that was done in my dialplan.
Now, it would be nice, if I can signaling on the phone which number will be
called, so that, for example, if I receive a call for +493511111111 I get a
message on the display or the phone ring with a particular tone, and if I
receive a call for +493512222222 the phone write something other on the
display or ring with another tone.

Is it possible? Maybe it depends from phone... I use a Thomson ST2022.

I don't know your phones, but there are multiple ways to achieve that. By far the easiest method
is to work with multiple SIP identities. You can adjust quite a few parameters, like display
text, ring tone, timings, forwarding ....

While you are busy with this, you can add additional accounts that operate as intercoms (baby
monitors) so you don't have to wait for an answer. Interesting exercise, but might disturb peace
in the house.

If your phone supports only a single identity, then you have to adjust caller ids, etc with
Asterisk.

jg

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gboelter at gmail.com
Guest

Posted: Sun May 31, 2015 3:43 am Post subject: [asterisk-users] Signaling incoming call

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Hash: SHA256

On 05/31/2015 02:31 PM, Luca Bertoncello wrote:

Quote:

Hi list!

Now all works as expected, at least in the simulation I did with
AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
changes my ISDN to VoIP...

Don't worry, Asterisk works very well with Deutsche Telekom and there
new ip-based connections ...

- --
DavaoSOFT, the home of ERPel
ERPel, das deutsche Warenwirtschaftssystem fuer LINUX
http://www.davaosoft.com
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--
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lucabert at lucabert.de
Guest

Posted: Sun May 31, 2015 3:59 am Post subject: [asterisk-users] Signaling incoming call

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Hash: SHA1

Guenther Boelter <gboelter@gmail.com> schrieb:

Quote:

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Hash: SHA256

On 05/31/2015 02:31 PM, Luca Bertoncello wrote:

Quote:

Hi list!

Now all works as expected, at least in the simulation I did with
AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
changes my ISDN to VoIP...

Don't worry, Asterisk works very well with Deutsche Telekom and there
new ip-based connections ...

That's a good news...
Currenty I configured my sip.conf so:

register => 00493511111111:MYSECRET@pbxluca/00493511111111
register => 00493512222222:MYSECRET@pbxfax/00493512222222
register => 00493513333333:MYSECRET@pbxanika/00493513333333
register => 4444444444:MYVERYSECRET@messagenet/4444444444

[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxanika]
type=peer
defaultuser=00493513333333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493513333333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[messagenet]
type=peer
defaultuser=4444444444
secret=MYVERYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=4444444444
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=60

Am I right if I say, that I just have to change "defaultuser", "host",
"secret", "outboundproxy" and "fromdomain" with the data from Telekom and it
works?

I thinks, it should be:

defaultuser=00493513333333
secret= MYSECRET
host=tel.t-online.de
context=anika_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=00493513333333
fromdomain=tel.t-online.de

I'm not sure, where I should write my Login (from my DSL-Line)...
I see this page (in German):

http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72

Could you please help me?

Thanks
Luca Bertoncello
(lucabert@lucabert.de)
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--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

sebastian_ml at gmx.net
Guest

Posted: Sun May 31, 2015 6:26 am Post subject: [asterisk-users] Signaling incoming call

Am 31. Mai 2015 10:58:56 MESZ, schrieb Luca Bertoncello <lucabert@lucabert.de>:

Quote:

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Guenther Boelter <gboelter@gmail.com> schrieb:

Quote:

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

On 05/31/2015 02:31 PM, Luca Bertoncello wrote:

Quote:

Hi list!

Now all works as expected, at least in the simulation I did with
AsteriskNOW. Hopefully it will work later, when Deutsche Telekom
changes my ISDN to VoIP...

Don't worry, Asterisk works very well with Deutsche Telekom and there
new ip-based connections ...

That's a good news...
Currenty I configured my sip.conf so:

register => 00493511111111:MYSECRET@pbxluca/00493511111111
register => 00493512222222:MYSECRET@pbxfax/00493512222222
register => 00493513333333:MYSECRET@pbxanika/00493513333333
register => 4444444444:MYVERYSECRET@messagenet/4444444444

[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxanika]
type=peer
defaultuser=00493513333333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493513333333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[messagenet]
type=peer
defaultuser=4444444444
secret=MYVERYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=4444444444
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=60

Am I right if I say, that I just have to change "defaultuser", "host",
"secret", "outboundproxy" and "fromdomain" with the data from Telekom
and it
works?

I thinks, it should be:

defaultuser=00493513333333
secret= MYSECRET
host=tel.t-online.de
context=anika_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=00493513333333
fromdomain=tel.t-online.de

I'm not sure, where I should write my Login (from my DSL-Line)...
I see this page (in German):

http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72

Could you please help me?

Thanks
Luca Bertoncello
(lucabert@lucabert.de)
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Hi Luca,

I had a discussion recently regarding Asterisk and your provider. The result you can basically find in this message: http://lists.digium.com/pipermail/asterisk-users/2015-April/286353.html

Regards,
Sebastian

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk.org at sedwar...
Guest

Posted: Sun May 31, 2015 2:25 pm Post subject: [asterisk-users] Signaling incoming call

On Sun, 31 May 2015, Luca Bertoncello wrote:

Quote:

register => 00493511111111:MYSECRET@pbxluca/00493511111111
register => 00493512222222:MYSECRET@pbxfax/00493512222222
register => 00493513333333:MYSECRET@pbxanika/00493513333333
register => 4444444444:MYVERYSECRET@messagenet/4444444444

[pbxluca]
type=peer
defaultuser=00493511111111
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493511111111
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[pbxfax]
type=peer
defaultuser=00493512222222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=00493512222222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
qualify=yes
qualifyfreq=600

[snip]

Just a few 'stylistic' and 'ease of maintenance' suggestions:

1) Group the 'register' with the stanza.

2) Add a bit of whitespace to increase 'readability.'

3) Sort the parameters to make it easier to maintain.

4) Move 'common' settings to 'general.'

Thus, I'd write your sip.conf as:

[general]
canreinvite = no
context = luca_incoming
dtmfmode = rfc2833
insecure = invite
outboundproxy = 172.16.34.132
port = 5060
qualify = yes
qualifyfreq = 600
usereqphone = yes

[general](+)
register = 00493511111111:MYSECRET@pbxluca/00493511111111
[pbxluca]
defaultuser = 00493511111111
fromdomain = 172.16.34.132
fromuser = 00493511111111
host = 172.16.34.132
secret = MYSECRET
type = peer

[general](+)
register = 00493512222222:MYSECRET@pbxfax/00493512222222
[pbxfax]
defaultuser = 00493512222222
fromdomain = 172.16.34.132
fromuser = 00493512222222
host = 172.16.34.132
secret = MYSECRET
type = peer

5) I'd try and move more of the common settings to general, but these were the ones listed on voip-info.org.

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

--
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asterisk.org at sedwar...
Guest

Posted: Sun May 31, 2015 5:13 pm Post subject: [asterisk-users] Signaling incoming call

On Sun, 31 May 2015, Luca Bertoncello wrote:

Quote:

Now, it would be nice, if I can signaling on the phone which number will
be called, so that, for example, if I receive a call for +493511111111 I
get a message on the display or the phone ring with a particular tone,
and if I receive a call for +493512222222 the phone write something
other on the display or ring with another tone.

Is it possible? Maybe it depends from phone... I use a Thomson ST2022.

You can fiddle with the caller ID to change what is displayed on the
phone.

You can fiddle with the ring tone by phone specific configuration and
phone specific SIP headers (sipaddheader(Alert-Info: ...)).

These seem relevant:

http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the
discussion looks relevant as well).

http://www.asteriskguru.com/tutorials/thomson_st2030.html

http://www.freepbx.org/support/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular-

--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello

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lucabert at lucabert.de
Guest

Posted: Mon Jun 01, 2015 12:28 am Post subject: [asterisk-users] Signaling incoming call

Steve Edwards <asterisk.org@sedwards.com> schrieb:

Quote:

Thank you very much!

I'll try it and report to the list.

Regards
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

lucabert at lucabert.de
Guest

Posted: Mon Jun 01, 2015 2:04 pm Post subject: [asterisk-users] Signaling incoming call

Steve Edwards <asterisk.org@sedwards.com> schrieb:

Quote:

You can fiddle with the ring tone by phone specific configuration and
phone specific SIP headers (sipaddheader(Alert-Info: ...)).

These seem relevant:

http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the
discussion looks relevant as well).

http://www.asteriskguru.com/tutorials/thomson_st2030.html

http://www.freepbx.org/support/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular-

Hi Steve!

Thank you very much!
It seems to run!

I wrote that:

exten => _00493513333333,n,Set(__ALERT_INFO=Bellcore-r3)
exten => _00493513333333,n,SIPAddHeader("Alert-Info:<http://www.notused.com>\;info=alert-external\;x-line-id=0")

and the phone rings with another melody.
Very curious is, that if I don't write BOTH lines, it does not run...

And, unfortunately, I just have two melody: the "normal" and this one, but it
is better than nothing!
Now, if it will be possible to add a text on the display, it will be perfect,
but I didn't found any option for that...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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kevin.larsen at pionee...
Guest

Posted: Mon Jun 01, 2015 2:08 pm Post subject: [asterisk-users] Signaling incoming call

Quote:

Hi Steve!

Thank you very much!
It seems to run!

I wrote that:

exten => _00493513333333,n,Set(__ALERT_INFO=Bellcore-r3)
exten => _00493513333333,n,SIPAddHeader("Alert-Info:<http://www.notused.com

Quote:

\;info=alert-external\;x-line-id=0")

and the phone rings with another melody.
Very curious is, that if I don't write BOTH lines, it does not run...

And, unfortunately, I just have two melody: the "normal" and this one, but it
is better than nothing!
Now, if it will be possible to add a text on the display, it will be perfect,
but I didn't found any option for that...

Look into Set(CALLERID(name)) and Set(CALLERID(num)) to manipulate the caller id name and number that show up on the phone.

lucabert at lucabert.de
Guest

Posted: Mon Jun 01, 2015 2:11 pm Post subject: [asterisk-users] Signaling incoming call

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

Look into Set(CALLERID(name)) and Set(CALLERID(num)) to manipulate the
caller id name and number that show up on the phone.

Hi Kevin

thank you very much for the suggestion.
I think it's not the right way, since I'd like to display the right number
and the name from address book...
If I change it, I'll not get the right data on the display, isn't it?

Anyway, I'll try tomorrow...

Thanks
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
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lucabert at lucabert.de
Guest

Posted: Tue Jun 02, 2015 11:26 am Post subject: [asterisk-users] Signaling incoming call

Kevin Larsen <kevin.larsen@pioneerballoon.com> schrieb:

Quote:

Look into Set(CALLERID(name)) and Set(CALLERID(num)) to manipulate the
caller id name and number that show up on the phone.

Hi Kevin!

Thanks! It works!
I can set the name of the line with CALLERID(name) and see the caller number,
too.
And, it the number is in the address book, I see the name, too.

Perfect!

Regards
Luca Bertoncello
(lucabert@lucabert.de)

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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kevin.larsen at pionee...
Guest

Posted: Tue Jun 02, 2015 11:31 am Post subject: [asterisk-users] Signaling incoming call

Quote:

Hi Kevin!

Thanks! It works!
I can set the name of the line with CALLERID(name) and see the caller number,
too.
And, it the number is in the address book, I see the name, too.

Perfect!

Glad it worked for you. I usually leave the number untouched, but will manipulate the name to suite what I want. I have mulitple call queues, so for instance, for my helpdesk lines, I will do something like transform "Name" to "HD:Name" so that the person being called knows that the caller dialed the help desk number rather than their direct number. On people who work multiple queues, it is very handy so they can see at a glance what queue the caller is reaching.

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