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hoomanfazaeli at gmail...
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 Posted: Mon Aug 15, 2016 2:37 pm Post subject: [asterisk-users] SIP 603 response when call is not answered |
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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. |
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info at jorisengbers.nl
Guest
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| Posted: Tue Aug 16, 2016 2:40 am Post subject: [asterisk-users] SIP 603 response when call is not answered |
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Hooman Fazaeli writes:
| Quote: | 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
--
_____________________________________________________________________
-- 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
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http://lists.digium.com/mailman/listinfo/asterisk-users |
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hoomanfazaeli at gmail...
Guest
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| Posted: Wed Aug 17, 2016 8:29 am Post subject: [asterisk-users] SIP 603 response when call is not answered |
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On 2016-08-16 12:10, Joris Engbers wrote:
| Quote: | Hooman Fazaeli writes:
| Quote: | 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@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@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@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@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@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@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
--
_____________________________________________________________________
-- 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 |
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mjordan at digium.com
Guest
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| Posted: Wed Aug 17, 2016 3:01 pm Post subject: [asterisk-users] SIP 603 response when call is not answered |
|
|
On Wed, Aug 17, 2016 at 8:29 AM, Hooman Fazaeli <hoomanfazaeli@gmail.com> wrote:
| Quote: | On 2016-08-16 12:10, Joris Engbers wrote:
| Quote: |
Hooman Fazaeli writes:
| Quote: | 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@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@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@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@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@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@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'
|
While it's a bit harsh, there's nothing inherently wrong with
returning a 603 in this case - so I wouldn't say it's a bug. Asterisk
has decided that since it tried everything it could about the
extension, the person at the other end must not want the call, and
hence opts for the global 6xx response as opposed to a 4xx response.
If you want to return something else, then you can provide a cause
code to the Hangup application that maps to the SIP response you'd
like to return. A mapping table exists on the wiki here:
https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings
In your case, providing a 19 to the Hangup dialplan application should
convert that to a SIP cause of 480.
--
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
--
_____________________________________________________________________
-- 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 |
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