asterisk users - Aug 2016 - SIP 603 response when call is not answered

Hi

I have noticed that asterisk returns 'SIP 603' when the called party
does
not answer.

My test setup is simple: two SIP phones (extensions: 100 and 111)
registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds.
When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to
111 (expected) and a '603 Decline' response to 100 (unexpected &
misleading).
It seems to me that a'480 Temporarily unavailable' response is more
suitable in this situation.

Is this a normal behavior of asterisk or a bug?

Thanks.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160816/9b624ac5/attachment.html>

Hooman Fazaeli writes:
> Hi
>
> I have noticed that asterisk returns 'SIP 603' when the called
party does
> not answer.
>
> My test setup is simple: two SIP phones (extensions: 100 and 111)
> registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds.
> When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to
> 111 (expected) and a '603 Decline' response to 100 (unexpected
&
> misleading).
> It seems to me that a'480 Temporarily unavailable' response is more
> suitable in this situation.
>
> Is this a normal behavior of asterisk or a bug?
>
> Thanks.
That sounds like you are not doing a Hangup().

What is the dialplan that you are using?

-- 
Joris Engbers

On 2016-08-16 12:10, Joris Engbers wrote:
> Hooman Fazaeli writes:
>
>> Hi
>>
>> I have noticed that asterisk returns 'SIP 603' when the called
party does
>> not answer.
>>
>> My test setup is simple: two SIP phones (extensions: 100 and 111)
>> registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30
seconds.
>> When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL
request to
>> 111 (expected) and a '603 Decline' response to 100 (unexpected
&
>> misleading).
>> It seems to me that a'480 Temporarily unavailable' response is
more
>> suitable in this situation.
>>
>> Is this a normal behavior of asterisk or a bug?
>>
>> Thanks.
> That sounds like you are not doing a Hangup().
>
> What is the dialplan that you are using?
>
Hangup() is there. The dial plan is:

(I set dial timeout to 10s to speed up tests)

[phone-100]
exten => 111,1,Dial(SIP/111,10,tTo)
exten => 111,n,Hangup()

[phone-111]
exten => 100,1,Dial(SIP/100,10,tTo)
exten => 100,n,Hangup()

As can be seen in below log messages, asterisk correctly sets DIALSTATUS to
NOANSWER (line 7).
Line 18 shows that the hangupcause value has been set to 16
(AST_CAUSE_NORMAL_CLEARING) which
asterisk complains has no SIP equivalent and falls back to 603. The problem
seems to be
that hangupcause is set incorrectly in the first place.

...
<Dial() times out>
...
1 VERBOSE[-1]: app_dial.c:1633 in wait_for_answer:     -- Nobody picked up in
10000 ms
2 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
'SIP/111-00000003'
3 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/111-00000003, SIP
callid 1a8ef4ce3f4d8a513de4639916c28b15 at 192.168.1.17:5060
4 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter for
outgoing call
5 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state Ringing
(not UP)
...
6 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'CANCEL sip:'
onto UDP socket destined for 192.168.1.200:5062
7 DEBUG[-1]: app_dial.c:3033 in dial_exec_full: Exiting with
DIALSTATUS=NOANSWER.
8 DEBUG[-1]: pbx.c:4720 in pbx_extension_helper: Launching 'Hangup'
9 VERBOSE[-1]: pbx.c:4728 in pbx_extension_helper:     -- Executing [111 at
phone-100:2] Hangup("SIP/100-00000002", "") in new stack
10 DEBUG[-1]: pbx.c:5544 in __ast_pbx_run: Spawn extension (phone-100,111,2)
exited non-zero on 'SIP/100-00000002'
11 VERBOSE[-1]: pbx.c:5545 in __ast_pbx_run:   == Spawn extension
(SIP-PHONE-35145790056fd369709fb2, 111, 2) exited non-zero on
'SIP/100-00000002'
12 DEBUG[-1]: channel.c:2735 in ast_softhangup_nolock: Soft-Hanging up channel
'SIP/100-00000002'
13 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
'SIP/100-00000002'
14 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/100-00000002, SIP
callid 9eda334cf9584d408ccd6e14eae7143a
15 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter for
incoming call
16 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state Ring
(not UP)
17 DEBUG[-1]: res_rtp_asterisk.c:2604 in ast_rtp_remote_address_set: Setting
RTCP address on RTP instance '0x29dbf01c'
18 DEBUG[-1]: chan_sip.c:6484 in hangup_cause2sip: AST hangup cause 16 (no match
found in SIP)
19 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'SIP/2.0 603'
onto UDP socket destined for 192.168.1.30:52628
20 ...
21 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
checking channel drivers for SIP - 111
22 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
peer 111
23 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/111 -
state 1 (Not in use)
24 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111'
state '1'
25 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
checking channel drivers for SIP - 111
26 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
peer 111
27 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/111 -
state 1 (Not in use)
28 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111'
state '1'
29 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
checking channel drivers for SIP - 100
30 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
peer 100
31 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/100 -
state 5 (Unavailable)
32 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100'
state '5'
33 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
9eda334cf9584d408ccd6e14eae7143a (Checking From) --From tag
48505f8775334f429a54d48d5b095543 --To-tag as2ad3ad05
34 DEBUG[-1]: chan_sip.c:25954 in handle_incoming: **** Received ACK (6) -
Command in SIP ACK
35 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
'9eda334cf9584d408ccd6e14eae7143a' of Response 21834: Match Found
36 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
1a8ef4ce3f4d8a513de4639916c28b15 at 192.168.1.17:5060 (Checking To) --From tag
as212fb4c7 --To-tag 479449046
37 DEBUG[-1]: chan_sip.c:4200 in __sip_ack: Acked pending invite 102
38 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
'1a8ef4ce3f4d8a513de4639916c28b15 at 192.168.1.17:5060' of Request 102:
Match Found
39 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
checking channel drivers for SIP - 100
40 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
peer 100
41 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/100 -
state 5 (Unavailable)
42 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100'
state '5'
43 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
1a8ef4ce3f4d8a513de4639916c28b15 at 192.168.1.17:5060 (Checking To) --From tag
as212fb4c7 --To-tag 479449046
44 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
'1a8ef4ce3f4d8a513de4639916c28b15 at 192.168.1.17:5060' of Request 102:
Match Found
45 DEBUG[-1]: chan_sip.c:20747 in handle_response_invite: SIP response 487 to
standard invite
46 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'ACK sip:111'
onto UDP socket destined for 192.168.1.200:5062
47 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter for
outgoing call
48 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
checking channel drivers for SIP - 111
49 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
peer 111
50 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/111 -
state 1 (Not in use)
51 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111'
state '1'

Any ideas?













-- 
Best regards
Hooman Fazaeli