<div dir="ltr">Hi everyone,<div>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.</div><div><br></div><div>The issue I am having is when I dial the other endpoint I receive 488 not acceptable here.</div><div><br></div><div><div>The following is the debug taken:</div><div>#########################<br></div><div><div>[Apr 1 15:07:39] VERBOSE[30911][C-00000000] app_dial.c: Called PJSIP/receiver@dtmf_inband</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction state change is TX_MSG</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending request</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending request</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Source address <a href="http://127.0.0.1:5060">127.0.0.1:5060</a> matches identify 'receiver'</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Retrieved endpoint receiver</div><div>[Apr 1 15:07:39] DEBUG[30861] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21</div><div>[Apr 1 15:07:39] DEBUG[30861] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE, Response is 488 Not Acceptable Here</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction state change is TX_MSG</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Sending response</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Method is INVITE, Response is 488 Not Acceptable Here</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Destroying SIP session with endpoint receiver</div><div>[Apr 1 15:07:39] DEBUG[30861] taskprocessor.c: destroying taskprocessor '22c1a0ee-5085-4a2f-8fe9-e3786ef73fb9'</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Source of transaction state change is RX_MSG</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Received response</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Response is 488 Not Acceptable Here</div><div>[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</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Source address <a href="http://127.0.0.1:5060">127.0.0.1:5060</a> matches identify 'receiver'</div><div>[Apr 1 15:07:39] DEBUG[30911][C-00000000] channel.c: Hanging up channel 'PJSIP/dtmf_inband-00000000'</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Retrieved endpoint receiver</div><div>[Apr 1 15:07:39] VERBOSE[30911][C-00000000] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)</div><div>[Apr 1 15:07:39] DEBUG[30861] res_pjsip_session.c: Destroying SIP session with endpoint dtmf_inband</div><div>[Apr 1 15:07:39] DEBUG[30911][C-00000000] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.</div><div>[Apr 1 15:07:39] DEBUG[30911][C-00000000] pbx.c: Launching 'Hangup'</div></div><div>#######################################</div></div><div><br></div><div>The following is the test scenario:</div><div>######################################</div><div>testinfo:</div><div> summary: 'Tests the PJSIP auto dtmf option'</div><div> description: |</div><div> 'Tests that dtmf settings is detected and setup according to the capabilities of the peer when auto dtmf is set'</div><div><br></div><div>test-modules:</div><div> test-object:</div><div> config-section: test-object-config</div><div> typename: 'test_case.SimpleTestCase'</div><div> modules:</div><div> -</div><div> config-section: ami-config</div><div> typename: 'ami.AMIEventModule'</div><div><br></div><div><br></div><div>test-object-config:</div><div> spawn-after-hangup: True</div><div> test-iterations:</div><div> -</div><div> channel: 'Local/dtmf_inband@default'</div><div> context: 'default'</div><div> exten: 'senddtmf'</div><div> priority: '1'</div><div><br></div><div>ami-config:</div><div> -</div><div> type: 'headermatch'</div><div> conditions:</div><div> match:</div><div> Event: 'DTMFEnd'</div><div> Channel: 'PJSIP/receiver-.*'</div><div> requirements:</div><div> match:</div><div> Digit: '1'</div><div> count: '1'</div><div><br></div><div>properties:</div><div> minversion: '13.4.0'</div><div> dependencies:</div><div> - python: 'twisted'</div><div> - python: 'starpy'</div><div> - asterisk: 'app_dial'</div><div> - asterisk: 'app_echo'</div><div> - asterisk: 'func_callerid'</div><div> - asterisk: 'chan_pjsip'</div><div><div> - asterisk: 'res_pjsip'</div><div> - asterisk: 'res_pjsip_caller_id'</div><div> - asterisk: 'res_pjsip_endpoint_identifier_user'</div><div> - asterisk: 'res_pjsip_sdp_rtp'</div><div> - asterisk: 'res_pjsip_session'</div><div> tags:</div><div> - pjsip</div></div><div>#########################</div><div><br></div><div>The following is the pjsip.conf</div><div>#########################<br></div><div><div>[local-transport]</div><div>type=transport</div><div>bind=127.0.0.1</div><div>protocol=udp</div><div><br></div><div>[dtmf_inband]</div><div>type=endpoint</div><div>dtmf_mode=inband</div><div>aors=dtmf_inband</div><div><br></div><div>[dtmf_inband]</div><div>type=aor</div><div>contact=sip:127.0.0.1</div><div><br></div><div><br></div><div>[receiver]</div><div>type=endpoint</div><div>dtmf_mode=inband</div><div><br></div><div><br></div><div>[receiver]</div><div>type=identify</div><div>endpoint=receiver</div><div>match=127.0.0.1</div></div><div>#########################<br></div><div><br></div><div>The following is the extensions.conf</div><div>#########################<br></div><div><div>[default]</div><div>exten => senddtmf,1,NoOp("YARON Is HERE SENDDTMF")</div><div>same => n,SendDTMF(1)</div><div>same => n,Hangup()</div><div><br></div><div>exten => dtmf_inband,1,NoOp("YARON Is HERE DIAL")</div><div>same => n,Dial(PJSIP/receiver@dtmf_inband)</div><div>same => n,Hangup()</div><div><br></div><div>exten => receiver,1,NoOp("YARON Is HERE RECEIVER")</div><div>same => n,Read(var)</div><div>same => n,Hangup()</div></div><div>#########################<br></div><div><br></div><div><br></div><div><br></div><div>Any help would be appreciated</div><div><br></div><div>Yaron.</div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Mar 31, 2015 at 6:43 PM, Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">On Tue, Mar 31, 2015 at 10:29 AM, Richard Mudgett <<a href="mailto:rmudgett@digium.com">rmudgett@digium.com</a>> wrote:<br>
> Another thing that is important is that the sample configs must be<br>
> installed.<br>
> Many tests have some difficulty if this is not the case. For me it was<br>
> because<br>
> I had configurations defining the same endpoints with chan_sip and<br>
> chan_pjsip.<br>
> The conflicting configs caused crashes in tests that did not use SIP at all.<br>
<br>
</div></div>*Most* of that has been resolved now, thanks to the 'is this test<br>
using chan_sip or chan_pjsip' logic added by Kevin.<br>
<br>
But generally, yes, using 'make samples' - or having enough<br>
configuration installed to get Asterisk up and running - is needed.<br>
<div class="HOEnZb"><div class="h5"><br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Director of Technology<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
<br>
asterisk-dev mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>
</div></div></blockquote></div><br></div>