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@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@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@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@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 <mjor...@digium.com> wrote: > On Tue, Mar 31, 2015 at 10:29 AM, Richard Mudgett <rmudg...@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 -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev >
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