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Matthew Jordan

2015-Sep-21 15:01 UTC

[asterisk-users] Respond to an out of call SIP MESSAGE

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net>
wrote:> On Mon, 21 Sep 2015 06:48:52 +0000
> Emil Ohlsson <emo at svep.se> wrote:
>> [sip-im]
>> exten _X!, 1, NoOp(Got message)
>> exten _X!, n, Answer()
>> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...)
>> exten _X!, n, SendText(Message received)
>
> I am not an expert but perhaps you want this.
>
> [sip-im]
>   exten s,1,NoOp(Got message)
>     same,n,Answer()
>     same,n,Agi(agi://localhost/messagehandler.agi?...)
>     same,n,SendText(Message received)
>
> Replacing "exten _X!" with "same" is just a shortcut. 
I find that
> there are lots of places where spaces cause problems so I just remove
> them all for good measure.  Finally, I am not sure what the mechanism
> is here but if it is like a goto then I think that you want the 's'
> priority.
>
> Or, I totally don't know what I am talking about and my education will
> be advanced by the replies to this message.  :-)
>
If you want to send an out of call SIP MESSAGE request, you'll need to
use the MessageSend application:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend

SendText is used for sending text messages within a call. Since a SIP
channel is not servicing the out of call text message, you cannot use
it to send a SIP MESSAGE request back to whatever sent the original
SIP MESSAGE request.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

Emil Ohlsson

2015-Sep-22 06:53 UTC

head link

[asterisk-users] Respond to an out of call SIP MESSAGE

Yes, that is how the problem was solved before when the transport protocol for
messages was UDP, but TLS (TCP) this doesn't work. Partly because
MessageSend doesn't support SIPS and partly because it focuses on sending a
new message instead of reusing an existing stream. Using wireshark I can see
that the TLS stream is still alive as there is no tear down communication.

The "same,n," is a good shortcut to make the text more readable,
I'll add that but I don't think that is the cause as I can follow the
source code and see that the SendText command is executed but fails silently.

Thanks for the feedback,
Emil

________________________________________
From: Matthew Jordan <mjordan at digium.com>
Sent: Monday, September 21, 2015 5:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Emil Ohlsson
Subject: Re: [asterisk-users] Respond to an out of call SIP MESSAGE

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net>
wrote:> On Mon, 21 Sep 2015 06:48:52 +0000
> Emil Ohlsson <emo at svep.se> wrote:
>> [sip-im]
>> exten _X!, 1, NoOp(Got message)
>> exten _X!, n, Answer()
>> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...)
>> exten _X!, n, SendText(Message received)
>
> I am not an expert but perhaps you want this.
>
> [sip-im]
>   exten s,1,NoOp(Got message)
>     same,n,Answer()
>     same,n,Agi(agi://localhost/messagehandler.agi?...)
>     same,n,SendText(Message received)
>
> Replacing "exten _X!" with "same" is just a shortcut. 
I find that
> there are lots of places where spaces cause problems so I just remove
> them all for good measure.  Finally, I am not sure what the mechanism
> is here but if it is like a goto then I think that you want the 's'
> priority.
>
> Or, I totally don't know what I am talking about and my education will
> be advanced by the replies to this message.  :-)
>
If you want to send an out of call SIP MESSAGE request, you'll need to
use the MessageSend application:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend

SendText is used for sending text messages within a call. Since a SIP
channel is not servicing the out of call text message, you cannot use
it to send a SIP MESSAGE request back to whatever sent the original
SIP MESSAGE request.

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

Joshua Colp

2015-Sep-22 10:45 UTC

head link

[asterisk-users] Respond to an out of call SIP MESSAGE

On 15-09-22 03:53 AM, Emil Ohlsson wrote:> Yes, that is how the problem was solved before when the transport
> protocol for messages was UDP, but TLS (TCP) this doesn't work.
> Partly because MessageSend doesn't support SIPS and partly because it
> focuses on sending a new message instead of reusing an existing
> stream. Using wireshark I can see that the TLS stream is still alive
> as there is no tear down communication.
What is the complete console output as well as endpoint configuration? 
It should work fine, provided stuff is configured and used right.

-- 
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

Emil Ohlsson

2015-Sep-28 12:16 UTC

head link

[asterisk-users] Respond to an out of call SIP MESSAGE

Sorry for the delay here. For some reason the mail from Joshua Colp failed to
deliver to my mailbox.

So, anyway, I've set up a local scenario on my computer a PJSIP client and
Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below.
In this scenario I've used UDP, but I want a configuration that can be used
with any transport protocol.

I can see that the context runs correctly, but there is no message delivered to
the client. Am I approaching this the wrong way?

sip.conf:
[general]
context=public                  ; Default context for incoming calls. Defaults
to 'default'
allowoverlap=no                 ; Disable overlap dialing support. (Default is
yes)
realm=my_realm                  ; Realm for digest authentication
udpbindaddr=0.0.0.0             ; IP address to bind UDP listen socket to
(0.0.0.0 binds to all)
tcpenable=no                    ; Enable server for incoming TCP connections
(default is no)
tcpbindaddr=0.0.0.0             ; IP address for TCP server to bind to (0.0.0.0
binds to all interfaces)
transport=udp                   ; Set the default transports.  The order
determines the primary default transport.
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
accept_outofcall_message = yes  ; Disable this option to reject all MESSAGE
requests outside of a
outofcall_message_context = tutorial ; Context all out of dialog msgs are sent
to. When this

[authentication]

[alice]
type=friend
host=dynamic
username=alice
secret=TopSecret
nat=no
transport=udp
context=tutorial

extensions.conf:
[general]
static=yes
writeprotect=no
clearglobalvars=no

[tutorial]
exten => _X!,1,NoOp(Got message)
exten => _X!,n,Answer()
exten => _X!,n,SendText(Zup dawg)
exten => _X!,n,NoOp(Done with message)

And when I run the scenario I get the following log:
<--- SIP read from UDP:127.0.0.1:5059 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1:5059;rport;branch=z9hG4bKPjd629f029-a971-4c83-9920-bb02da26cc25
Max-Forwards: 70
From: <sip:alice at 127.0.0.1>;tag=4b251b50-2a30-498c-a7bd-9502271e8ac2
To: <sip:alice at 127.0.0.1>
Call-ID: be616b5b-a301-477e-a4bc-c011bcecd4f0
CSeq: 17321 REGISTER
Contact: <sip:alice at 127.0.0.1:5059;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 127.0.0.1:5059 (no NAT)
Sending to 127.0.0.1:5059 (no NAT)

<--- Transmitting (no NAT) to 127.0.0.1:5059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5059;branch=z9hG4bKPjd629f029-a971-4c83-9920-bb02da26cc25;received=127.0.0.1;rport=5059
From: <sip:alice at 127.0.0.1>;tag=4b251b50-2a30-498c-a7bd-9502271e8ac2
To: <sip:alice at 127.0.0.1>;tag=as62d4da5c
Call-ID: be616b5b-a301-477e-a4bc-c011bcecd4f0
CSeq: 17321 REGISTER
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="my_realm",
nonce="394fd8d9"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'be616b5b-a301-477e-a4bc-c011bcecd4f0' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:127.0.0.1:5059 --->
REGISTER sip:127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1:5059;rport;branch=z9hG4bKPj8905923d-2405-45dc-97c9-8abb213075c7
Max-Forwards: 70
From: <sip:alice at 127.0.0.1>;tag=4b251b50-2a30-498c-a7bd-9502271e8ac2
To: <sip:alice at 127.0.0.1>
Call-ID: be616b5b-a301-477e-a4bc-c011bcecd4f0
CSeq: 17322 REGISTER
Contact: <sip:alice at 127.0.0.1:5059;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Authorization: Digest username="alice", realm="my_realm",
nonce="394fd8d9", uri="sip:127.0.0.1:5060",
response="e72c84753c7768e0048c094d37ac6c70", algorithm=MD5
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 127.0.0.1:5059 (no NAT)

<--- Transmitting (no NAT) to 127.0.0.1:5059 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.0.1:5059;branch=z9hG4bKPj8905923d-2405-45dc-97c9-8abb213075c7;received=127.0.0.1;rport=5059
From: <sip:alice at 127.0.0.1>;tag=4b251b50-2a30-498c-a7bd-9502271e8ac2
To: <sip:alice at 127.0.0.1>;tag=as62d4da5c
Call-ID: be616b5b-a301-477e-a4bc-c011bcecd4f0
CSeq: 17322 REGISTER
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 300
Contact: <sip:alice at 127.0.0.1:5059;ob>;expires=300
Date: Mon, 28 Sep 2015 12:03:22 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'be616b5b-a301-477e-a4bc-c011bcecd4f0' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:127.0.0.1:5059 --->
MESSAGE sip:1234 at 127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1:5059;rport;branch=z9hG4bKPjb2135f5d-b7ba-43e6-a1d9-1f2729b2f1b2
Max-Forwards: 70
From: <sip:alice at 127.0.0.1>;tag=c12f77aa-d0d5-4517-8430-bf4406006397
To: <sip:1234 at 127.0.0.1>
Call-ID: 1b6ac040-4af5-4015-a85d-fb5c48831d6f
CSeq: 1266 MESSAGE
Accept: text/plain, application/im-iscomposing+xml
Content-Type: text/plain
Content-Length: 13

Hello, world!
<------------->
--- (10 headers 1 lines) ---
Sending to 127.0.0.1:5059 (no NAT)
Receiving message!
Found peer 'alice' for 'alice' from 127.0.0.1:5059

<--- Transmitting (no NAT) to 127.0.0.1:5059 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5059;branch=z9hG4bKPjb2135f5d-b7ba-43e6-a1d9-1f2729b2f1b2;received=127.0.0.1;rport=5059
From: <sip:alice at 127.0.0.1>;tag=c12f77aa-d0d5-4517-8430-bf4406006397
To: <sip:1234 at 127.0.0.1>;tag=as32a0932b
Call-ID: 1b6ac040-4af5-4015-a85d-fb5c48831d6f
CSeq: 1266 MESSAGE
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="my_realm",
nonce="0698f476"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'1b6ac040-4af5-4015-a85d-fb5c48831d6f' in 32000 ms (Method: MESSAGE)
Scheduling destruction of SIP dialog
'1b6ac040-4af5-4015-a85d-fb5c48831d6f' in 32000 ms (Method: MESSAGE)

<--- SIP read from UDP:127.0.0.1:5059 --->
MESSAGE sip:1234 at 127.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1:5059;rport;branch=z9hG4bKPj71f66ce8-9f0f-4841-aca8-ced79085ad65
Max-Forwards: 70
From: <sip:alice at 127.0.0.1>;tag=c12f77aa-d0d5-4517-8430-bf4406006397
To: <sip:1234 at 127.0.0.1>
Call-ID: 1b6ac040-4af5-4015-a85d-fb5c48831d6f
CSeq: 1267 MESSAGE
Accept: text/plain, application/im-iscomposing+xml
Authorization: Digest username="alice", realm="my_realm",
nonce="0698f476", uri="sip:1234 at 127.0.0.1:5060",
response="8d4ebff55b35d0363e4cd726c8f410c7", algorithm=MD5
Content-Type: text/plain
Content-Length: 13

Hello, world!
<------------->
--- (11 headers 1 lines) ---
Receiving message!
Found peer 'alice' for 'alice' from 127.0.0.1:5059
Looking for 1234 in tutorial (domain 127.0.0.1)

<--- Transmitting (no NAT) to 127.0.0.1:5059 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
127.0.0.1:5059;branch=z9hG4bKPj71f66ce8-9f0f-4841-aca8-ced79085ad65;received=127.0.0.1;rport=5059
From: <sip:alice at 127.0.0.1>;tag=c12f77aa-d0d5-4517-8430-bf4406006397
To: <sip:1234 at 127.0.0.1>;tag=as32a0932b
Call-ID: 1b6ac040-4af5-4015-a85d-fb5c48831d6f
CSeq: 1267 MESSAGE
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog
'1b6ac040-4af5-4015-a85d-fb5c48831d6f' in 32000 ms (Method: MESSAGE)
    -- Executing [1234 at tutorial:1] NoOp("Message/ast_msg_queue",
"Got message") in new stack
    -- Executing [1234 at tutorial:2] Answer("Message/ast_msg_queue",
"") in new stack
    -- Executing [1234 at tutorial:3]
SendText("Message/ast_msg_queue", "Zup dawg") in new stack
    -- Executing [1234 at tutorial:4] NoOp("Message/ast_msg_queue",
"Done with message") in new stack
    -- Auto fallthrough, channel 'Message/ast_msg_queue' status is
'UNKNOWN'

BR,
Emil

________________________________________
From: Emil Ohlsson
Sent: Tuesday, September 22, 2015 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Respond to an out of call SIP MESSAGE

Yes, that is how the problem was solved before when the transport protocol for
messages was UDP, but TLS (TCP) this doesn't work. Partly because
MessageSend doesn't support SIPS and partly because it focuses on sending a
new message instead of reusing an existing stream. Using wireshark I can see
that the TLS stream is still alive as there is no tear down communication.

The "same,n," is a good shortcut to make the text more readable,
I'll add that but I don't think that is the cause as I can follow the
source code and see that the SendText command is executed but fails silently.

Thanks for the feedback,
Emil

________________________________________
From: Matthew Jordan <mjordan at digium.com>
Sent: Monday, September 21, 2015 5:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Emil Ohlsson
Subject: Re: [asterisk-users] Respond to an out of call SIP MESSAGE

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net>
wrote:> On Mon, 21 Sep 2015 06:48:52 +0000
> Emil Ohlsson <emo at svep.se> wrote:
>> [sip-im]
>> exten _X!, 1, NoOp(Got message)
>> exten _X!, n, Answer()
>> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...)
>> exten _X!, n, SendText(Message received)
>
> I am not an expert but perhaps you want this.
>
> [sip-im]
>   exten s,1,NoOp(Got message)
>     same,n,Answer()
>     same,n,Agi(agi://localhost/messagehandler.agi?...)
>     same,n,SendText(Message received)
>
> Replacing "exten _X!" with "same" is just a shortcut. 
I find that
> there are lots of places where spaces cause problems so I just remove
> them all for good measure.  Finally, I am not sure what the mechanism
> is here but if it is like a goto then I think that you want the 's'
> priority.
>
> Or, I totally don't know what I am talking about and my education will
> be advanced by the replies to this message.  :-)
>
If you want to send an out of call SIP MESSAGE request, you'll need to
use the MessageSend application:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend

SendText is used for sending text messages within a call. Since a SIP
channel is not servicing the out of call text message, you cannot use
it to send a SIP MESSAGE request back to whatever sent the original
SIP MESSAGE request.

--
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

asterisk users - Sep 2015 - Respond to an out of call SIP MESSAGE

[asterisk-users] Respond to an out of call SIP MESSAGE

[asterisk-users] Respond to an out of call SIP MESSAGE

[asterisk-users] Respond to an out of call SIP MESSAGE

[asterisk-users] Respond to an out of call SIP MESSAGE