[asterisk-users] how do i enable call features??

2018-01-09 Thread Atux Atux
Hi. i am running asterisk 11 and i would like to have features access codes
in my system such as call waiting(all types) (enable/disable), call forward
(enable/disable) and DND. my dialplan is pretty simple and it is the
following

[DefaultPlan]exten =>
_XX,1,Dial(SIP/VoipGate/${EXTEN},120,Tt)exten =>
_XX,1,Busy()
exten => _4XX,2,Answer()exten =>
_4XX,3,VoiceMail(${EXTEN}@Office,su)exten => _4XX,4,HangUp()exten =>
_4XX,102,Answer()exten => _4XX,103,VoiceMail(${EXTEN}@Office,sb)exten
=> _4XX,104,HangUp()

i would like to enable/disable call waiting by typing eg. *70/*71
DND for the extension *72 enable, *73 to disable.

Regarding call waiting, at the moment it is disabled (default value). Now
if an extension is busy, a busy message is send back to the caller. I would
like have the following behavior:
-in the event were the extension is busy, then send a message indication to
the extension and the caller to hear from the SIP provider the default
early media for call waiting due to busy. Then after some period of time eg
30 secs send busy.
-in the event where the extension is busy, send the early media to the
caller and waiting indication to the extension. If the extension decides to
get the call then get the 2nd call and send the 1st to hear moh.

My phones are mainly softphones (zoiper), a few IP phones and 2 SPA3000 for
analog devices.

could someone help me please with this task, please?
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Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

2018-01-09 Thread Joshua Colp
On Tue, Jan 9, 2018, at 10:01 AM, Benoit Panizzon wrote:
> Hi Jushua
> 
> > The rtp_ipv6 option is not needed, in current versions things will
> > automatically be updated to reflect the signaling. Remove it and give
> > it a try. The option itself actually had the bug that you are seeing.
> 
> Ok, commented out rtp_ipv6 in the config and did try again:
> 
> IPv6 Registered client.
> 
> c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80
> 
> Reply from *
> 
> c=IN IP6 2001:4060:dead:beef::1
> 
> IPv4 registered client:
> 
> c=IN IP4 157.161.4.172
> 
> Reply from *
> 
> c=IN IP4 157.161.57.1
> 
> Perfect! It didn't occur to me to completely comment out that option as
> I believed it was needed for rtp to work over ipv6.
> 
> Thank you for that exceptional quick help.

It used to be required but as part of Asterisk 14 work was done in DNS land 
(failover to different targets, including between IPv6 and IPv4) and based on 
discussions I had with other people at SIPit I made it automatic so that media 
family = signaling family. To keep things better in line and to provide a 
better experience the change was also done in Asterisk 13.

-- 
Joshua Colp
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 rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

2018-01-09 Thread Benoit Panizzon
Hi Jushua

> The rtp_ipv6 option is not needed, in current versions things will
> automatically be updated to reflect the signaling. Remove it and give
> it a try. The option itself actually had the bug that you are seeing.

Ok, commented out rtp_ipv6 in the config and did try again:

IPv6 Registered client.

c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80

Reply from *

c=IN IP6 2001:4060:dead:beef::1

IPv4 registered client:

c=IN IP4 157.161.4.172

Reply from *

c=IN IP4 157.161.57.1

Perfect! It didn't occur to me to completely comment out that option as
I believed it was needed for rtp to work over ipv6.

Thank you for that exceptional quick help.

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

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Re: [asterisk-users] pjsip rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

2018-01-09 Thread Joshua Colp
On Tue, Jan 9, 2018, at 9:48 AM, Benoit Panizzon wrote:
> Dear List
> 
> I fear I stumbled over a bug in asterisk 13.14.1.
> 
> My 'phones' are roaming around, sometimes some are connecting from ipv6
> enabled networks, another time they are not.
> 
> If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat
> problems.
> 
> I have not specified a transport in the endpoint section, so that the
> appropriate transport which corresponds to the registration can be used.
> 
> Now I have noticed, if an phone is registered from an ipv4 only
> endpoint and is performing an outgoing call, my asterisk server is
> answering with an IP4 RTP IPv6 address:

The rtp_ipv6 option is not needed, in current versions things will 
automatically be updated to reflect the signaling. Remove it and give it a try. 
The option itself actually had the bug that you are seeing.

-- 
Joshua Colp
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 rtp_ipv6=yes but endpoint registered via ipv4 (IP4 contact infor)

2018-01-09 Thread Benoit Panizzon
Dear List

I fear I stumbled over a bug in asterisk 13.14.1.

My 'phones' are roaming around, sometimes some are connecting from ipv6
enabled networks, another time they are not.

If a connection is ipv6 I would prefer to use ipv6 to avoid ipv4-nat
problems.

I have not specified a transport in the endpoint section, so that the
appropriate transport which corresponds to the registration can be used.

Now I have noticed, if an phone is registered from an ipv4 only
endpoint and is performing an outgoing call, my asterisk server is
answering with an IP4 RTP IPv6 address:

Example:

<--- Received SIP request (1235 bytes) from UDP:157.161.4.172:5060 --->
[...]
v=0
o=MxSIP 0 20 IN IP4 157.161.4.172
s=SIP Call
c=IN IP4 157.161.4.172
t=0 0
m=audio 6018 RTP/AVP 9 8 97 91 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 CLEARMODE/8000
a=rtpmap:91 X-CLEAR-CHANNEL/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv

<--- Transmitting SIP response (914 bytes) to UDP:157.161.4.172:5060
--->
[...]
v=0
o=- 0 22 IN IP4 2001:4060:dead:beef::1
s=Asterisk
c=IN IP4 2001:4060:dead:beef::1
t=0 0
m=audio 16172 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


Errr... how should my client @ 157.161.4.172 send udp to
2001:4060:dead:beef::1?

Also when I compare with a real IPv6 client I notice from the client:

c=IN IP6 2001:4060:1:4133:204:13ff:fe30:2a80

from the Asterisk:

c=IN IP4 2001:4060:dead:beef::1

Shouldn't that also be IP6 from the asterisk?

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

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Re: [asterisk-users] PJSIP: identify endpoint by authentication username?

2018-01-09 Thread Benoit Panizzon
Hi George

> [global]
> endpoint_identifier_order = auth_username,username,ip,anonymous
> 
> [endpoint_x]
> identify_by = auth_username

Thank you, I missed that config option, works perfectly!

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
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Re: [asterisk-users] PJSIP: identify endpoint by authentication username?

2018-01-09 Thread George Joseph
On Tue, Jan 9, 2018 at 5:38 AM, Benoit Panizzon 
wrote:

> Dear fellow list readers
>
> This is the situation:
>
> ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP
>
> The Patton GW resides on a dynamic IP address, so I cannot really use
> match=ip in the identify section.
>
> The Patton does not send a line parameter.
>
> The ISDN Devices behind the patton have different MSN and should be
> able to send them in the From: Header, so the default endpoint
> identification mechanism which matches the From username with the
> endplaint fails.
>
> So what are the options to solve that issue?
>
> I see the asterisk sending out a challenge and getting a proper reply
> from the patton, but then stills complains about the endpoint not
> matching.
>
> According to the manual there is no
>
> type=identify
> match=authentication_username
>

There is no need for a separate "identify" object in this case.  In the
pjsip.conf "global" section set "endpoint_identifier_order" to include
"auth_username" and in each endpoint's section set "identify_by" to include
"auth_username".

[global]
endpoint_identifier_order = auth_username,username,ip,anonymous

[endpoint_x]
identify_by = auth_username





>
> or similar.
>
> Mit freundlichen Grüssen
>
> -Benoît Panizzon-
> --
> I m p r o W a r e   A G-Leiter Commerce Kunden
> __
>
> Zurlindenstrasse 29 Tel  +41 61 826 93 00
> CH-4133 PrattelnFax  +41 61 826 93 01
> Schweiz Web  http://www.imp.ch
> __
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users




-- 
George Joseph
Digium, Inc. | 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: identify endpoint by authentication username?

2018-01-09 Thread Benoit Panizzon
Dear fellow list readers

This is the situation:

ISDN Devices => Patton ISDN to SIP GW => Asterisk PJSIP

The Patton GW resides on a dynamic IP address, so I cannot really use
match=ip in the identify section.

The Patton does not send a line parameter.

The ISDN Devices behind the patton have different MSN and should be
able to send them in the From: Header, so the default endpoint
identification mechanism which matches the From username with the
endplaint fails.

So what are the options to solve that issue?

I see the asterisk sending out a challenge and getting a proper reply
from the patton, but then stills complains about the endpoint not
matching.

According to the manual there is no

type=identify
match=authentication_username

or similar.

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

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