[asterisk-dev] running pjsip testsuite
Yaron Nachum
nachum.yaron at gmail.com
Wed Apr 1 07:20:17 CDT 2015
Hi everyone,
I am still having problems with the testsuite. I made a simple scenario
that originates a call from the ami to a local channel, an then dials
through a PJSIP endpoint to another PJSIP endpoint.
The issue I am having is when I dial the other endpoint I receive 488 not
acceptable here.
The following is the debug taken:
#########################
[Apr 1 15:07:39] VERBOSE[30911][C-00000000] app_dial.c: Called
PJSIP/receiver at dtmf_inband
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction
state change is TX_MSG
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending request
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending request
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Source
address 127.0.0.1:5060 matches identify 'receiver'
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c:
Retrieved endpoint receiver
[Apr 1 15:07:39] DEBUG[30861] dsp.c: Setup tone 1100 Hz, 500 ms,
block_size=160, hits_required=21
[Apr 1 15:07:39] DEBUG[30861] dsp.c: Setup tone 2100 Hz, 2600 ms,
block_size=160, hits_required=116
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE,
Response is 488 Not Acceptable Here
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction
state change is TX_MSG
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending response
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE,
Response is 488 Not Acceptable Here
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Destroying SIP session
with endpoint receiver
[Apr 1 15:07:39] DEBUG[30861] taskprocessor.c: destroying taskprocessor
'22c1a0ee-5085-4a2f-8fe9-e3786ef73fb9'
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction
state change is RX_MSG
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Received response
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Response is 488 Not
Acceptable Here
[Apr 1 15:07:39] DEBUG[30848] cdr.c: Finalized CDR for
Local/dtmf_inband at default-00000000;2 - start 1427890059.401763 answer
0.000000 end 1427890059.407073 dispo NO ANSWER
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Source
address 127.0.0.1:5060 matches identify 'receiver'
[Apr 1 15:07:39] DEBUG[30911][C-00000000] channel.c: Hanging up channel
'PJSIP/dtmf_inband-00000000'
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c:
Retrieved endpoint receiver
[Apr 1 15:07:39] VERBOSE[30911][C-00000000] app_dial.c: Everyone is
busy/congested at this time (1:0/0/1)
[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Destroying SIP session
with endpoint dtmf_inband
[Apr 1 15:07:39] DEBUG[30911][C-00000000] app_dial.c: Exiting with
DIALSTATUS=CHANUNAVAIL.
[Apr 1 15:07:39] DEBUG[30911][C-00000000] pbx.c: Launching 'Hangup'
#######################################
The following is the test scenario:
######################################
testinfo:
summary: 'Tests the PJSIP auto dtmf option'
description: |
'Tests that dtmf settings is detected and setup according to the
capabilities of the peer when auto dtmf is set'
test-modules:
test-object:
config-section: test-object-config
typename: 'test_case.SimpleTestCase'
modules:
-
config-section: ami-config
typename: 'ami.AMIEventModule'
test-object-config:
spawn-after-hangup: True
test-iterations:
-
channel: 'Local/dtmf_inband at default'
context: 'default'
exten: 'senddtmf'
priority: '1'
ami-config:
-
type: 'headermatch'
conditions:
match:
Event: 'DTMFEnd'
Channel: 'PJSIP/receiver-.*'
requirements:
match:
Digit: '1'
count: '1'
properties:
minversion: '13.4.0'
dependencies:
- python: 'twisted'
- python: 'starpy'
- asterisk: 'app_dial'
- asterisk: 'app_echo'
- asterisk: 'func_callerid'
- asterisk: 'chan_pjsip'
- asterisk: 'res_pjsip'
- asterisk: 'res_pjsip_caller_id'
- asterisk: 'res_pjsip_endpoint_identifier_user'
- asterisk: 'res_pjsip_sdp_rtp'
- asterisk: 'res_pjsip_session'
tags:
- pjsip
#########################
The following is the pjsip.conf
#########################
[local-transport]
type=transport
bind=127.0.0.1
protocol=udp
[dtmf_inband]
type=endpoint
dtmf_mode=inband
aors=dtmf_inband
[dtmf_inband]
type=aor
contact=sip:127.0.0.1
[receiver]
type=endpoint
dtmf_mode=inband
[receiver]
type=identify
endpoint=receiver
match=127.0.0.1
#########################
The following is the extensions.conf
#########################
[default]
exten => senddtmf,1,NoOp("YARON Is HERE SENDDTMF")
same => n,SendDTMF(1)
same => n,Hangup()
exten => dtmf_inband,1,NoOp("YARON Is HERE DIAL")
same => n,Dial(PJSIP/receiver at dtmf_inband)
same => n,Hangup()
exten => receiver,1,NoOp("YARON Is HERE RECEIVER")
same => n,Read(var)
same => n,Hangup()
#########################
Any help would be appreciated
Yaron.
On Tue, Mar 31, 2015 at 6:43 PM, Matthew Jordan <mjordan at digium.com> wrote:
> On Tue, Mar 31, 2015 at 10:29 AM, Richard Mudgett <rmudgett at digium.com>
> wrote:
> > Another thing that is important is that the sample configs must be
> > installed.
> > Many tests have some difficulty if this is not the case. For me it was
> > because
> > I had configurations defining the same endpoints with chan_sip and
> > chan_pjsip.
> > The conflicting configs caused crashes in tests that did not use SIP at
> all.
>
> *Most* of that has been resolved now, thanks to the 'is this test
> using chan_sip or chan_pjsip' logic added by Kevin.
>
> But generally, yes, using 'make samples' - or having enough
> configuration installed to get Asterisk up and running - is needed.
>
> --
> 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
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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