subject:"\[asterisk\-users\] SIP 603 response when call is not answered"

Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Matthew Jordan

On Wed, Aug 17, 2016 at 8:29 AM, Hooman Fazaeli  wrote:
> 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.
>
> ...
> 
> ...
> 1 VERBOSE[-1]: app_dial.c:1633 in wait_for_answer: -- Nobody picked up
> in 1 ms
> 2 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
> 'SIP/111-0003'
> 3 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/111-0003,
> 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/100-0002", "") in new stack
> 10 DEBUG[-1]: pbx.c:5544 in __ast_pbx_run: Spawn extension (phone-100,111,2)
> exited non-zero on 'SIP/100-0002'
> 11 VERBOSE[-1]: pbx.c:5545 in __ast_pbx_run:   == Spawn extension
> (SIP-PHONE-35145790056fd369709fb2, 111, 2) exited non-zero on
> 'SIP/100-0002'
> 12 DEBUG[-1]: channel.c:2735 in ast_softhangup_nolock: Soft-Hanging up
> channel 'SIP/100-0002'
> 13 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
> 'SIP/100-0002'
> 14 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/100-0002,
> 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

Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Hooman Fazaeli


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.

...

...
1 VERBOSE[-1]: app_dial.c:1633 in wait_for_answer: -- Nobody picked up in 
1 ms
2 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel 'SIP/111-0003'
3 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/111-0003, 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/100-0002", "") in new stack
10 DEBUG[-1]: pbx.c:5544 in __ast_pbx_run: Spawn extension (phone-100,111,2) 
exited non-zero on 'SIP/100-0002'
11 VERBOSE[-1]: pbx.c:5545 in __ast_pbx_run:   == Spawn extension 
(SIP-PHONE-35145790056fd369709fb2, 111, 2) exited non-zero on 'SIP/100-0002'
12 DEBUG[-1]: channel.c:2735 in ast_softhangup_nolock: Soft-Hanging up channel 
'SIP/100-0002'
13 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel 
'SIP/100-0002'
14 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/100-0002, 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

Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-16 Thread Joris Engbers


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

-- 
_
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[asterisk-users] SIP 603 response when call is not answered

2016-08-15 Thread Hooman Fazaeli

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.
-- 
_
-- 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

Re: [asterisk-users] SIP 603 response when call is not answered

Re: [asterisk-users] SIP 603 response when call is not answered

Re: [asterisk-users] SIP 603 response when call is not answered

[asterisk-users] SIP 603 response when call is not answered

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