Re: [asterisk-dev] running pjsip testsuite
Hi everyone,
I ran the test manually. Just setup a single endpoint and using AMI I
originanted a call to an extension which dials to another extension and
send DTMF sequence using SendDTMF application.
When I setup the endpoint with rfc4733 the dtmf is identified, but when I
setup the endpoint with inband it is not identified. Using rtp debug I see
that the rtp is sent and received.
I did the same scenario with regular sip channel and the same happened.
If anyone has a clue please get back to me.
I will try to make the test with sipp.
Yaron
On Wed, Apr 1, 2015 at 6:34 PM, Yaron Nachum nachum.ya...@gmail.com wrote:
Hi Everyone,
Sorry for all the questions.
Well I managed to understand the 488 issue - I had to add some codec
capabilities. Now the test works but only if I setup the dtmfmode to
rfc4733. If I set it to inband it fails - the Read on the receiver side
doesn't receive DTMF.
The following is the 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: 'PJSIP/dtmf_inband@dtmf_inband'
context: 'default'
exten: 'senddtmf'
priority: '1'
ami-config:
-
type: 'headermatch'
conditions:
match:
Event: 'DTMFEnd'
Channel: 'PJSIP/receiver-.*'
Exten: '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 extensions.conf:
[default]
exten = senddtmf,1,NoOp(YARON Is HERE SENDDTMF
dtmfmode=${PJSIP_ENDPOINT(dtmf_inband,dtmf_mode)})
same = n,Dumpchan()
;same = n,SendDTMF(1)
same = n,Wait(5)
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 dtmfmode =
${PJSIP_ENDPOINT(receiver,dtmf_mode)})
same = n,Dumpchan()
same = n,Answer()
same = n,Read(var,,1,,1,4)
same = n,NoOp(YARON Is HERE var=${var})
same = n,Hangup()
--
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Re: [asterisk-dev] running pjsip testsuite
On Thu, Apr 2, 2015 at 3:25 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi everyone, I ran the test manually. Just setup a single endpoint and using AMI I originanted a call to an extension which dials to another extension and send DTMF sequence using SendDTMF application. When I setup the endpoint with rfc4733 the dtmf is identified, but when I setup the endpoint with inband it is not identified. Using rtp debug I see that the rtp is sent and received. I did the same scenario with regular sip channel and the same happened. If anyone has a clue please get back to me. I will try to make the test with sipp. Hey Yaron - Can you attach a DEBUG log snippet from the Asterisk instance/channel sending DTMF inband? In particular, the part where it does the negotiation, along with sending the DTMF digit. Thanks! -- 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
Re: [asterisk-dev] running pjsip testsuite
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-] 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-;2 - start 1427890059.401763 answer 0.00 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-] channel.c: Hanging up channel 'PJSIP/dtmf_inband-' [Apr 1 15:07:39] DEBUG[30861] res_pjsip_endpoint_identifier_ip.c: Retrieved endpoint receiver [Apr 1 15:07:39] VERBOSE[30911][C-] 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-] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. [Apr 1 15:07:39] DEBUG[30911][C-] 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
Re: [asterisk-dev] running pjsip testsuite
Hi Everyone,
Sorry for all the questions.
Well I managed to understand the 488 issue - I had to add some codec
capabilities. Now the test works but only if I setup the dtmfmode to
rfc4733. If I set it to inband it fails - the Read on the receiver side
doesn't receive DTMF.
The following is the 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: 'PJSIP/dtmf_inband@dtmf_inband'
context: 'default'
exten: 'senddtmf'
priority: '1'
ami-config:
-
type: 'headermatch'
conditions:
match:
Event: 'DTMFEnd'
Channel: 'PJSIP/receiver-.*'
Exten: '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 extensions.conf:
[default]
exten = senddtmf,1,NoOp(YARON Is HERE SENDDTMF
dtmfmode=${PJSIP_ENDPOINT(dtmf_inband,dtmf_mode)})
same = n,Dumpchan()
;same = n,SendDTMF(1)
same = n,Wait(5)
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 dtmfmode =
${PJSIP_ENDPOINT(receiver,dtmf_mode)})
same = n,Dumpchan()
same = n,Answer()
same = n,Read(var,,1,,1,4)
same = n,NoOp(YARON Is HERE var=${var})
same = n,Hangup()
--
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Re: [asterisk-dev] running pjsip testsuite
On Tue, Mar 31, 2015 at 8:04 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi everyone, I am trying to add some tests for the PJSIP auto-dtmf support . Before I start I just wanted to run some of the existing tests in order to understand the process. Whenever I try to run a test from the pjsip tests I get - --- -- Dependency: res_pjsip - False. The following output was generated when I tried to run the rpid_immediate test: [root@stnrd5652 testsuite]# ./runtests.py -t tests/channels/pjsip/rpid_immediate/ Running tests for Asterisk SVN-trunk-r431522M ... Tests to run: 0, Maximum test inactivity time: -1 sec. -- Cannot run test 'tests/channels/pjsip/rpid_immediate' --- -- Minimum Version: 13.4.0 (True) --- -- Maximum Version: (True) --- -- Tags: ['pjsip'] --- -- Dependency: twisted - True --- -- Dependency: starpy - True --- -- Dependency: app_dial - True --- -- Dependency: app_echo - True --- -- Dependency: func_callerid - True --- -- Dependency: chan_pjsip - False --- -- Dependency: res_pjsip - False --- -- Dependency: res_pjsip_caller_id - False --- -- Dependency: res_pjsip_endpoint_identifier_user - False --- -- Dependency: res_pjsip_sdp_rtp - False --- -- Dependency: res_pjsip_session - False ?xml version=1.0 encoding=utf-8? testsuite errors=0 failures=0 name=AsteriskTestSuite tests=0 time=0.00/ Am I doing anything stupid? Congratulations on getting this far! Looks like you've got most of the dependencies worked out in the testsuite. The dependency checking for 'res_pjsip' is looking at what modules you have installed on the system. Double check that your Asterisk installation did detect pjproject, and that it built and installed the res_pjsip* modules. You can check this in menuselect. Matt -- 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
Re: [asterisk-dev] running pjsip testsuite
The following is the output [root@stnrd5652 testsuite]# ls /usr/lib/asterisk/modules | grep pjsip chan_pjsip.so func_pjsip_aor.so func_pjsip_contact.so func_pjsip_endpoint.so res_hep_pjsip.so res_pjsip_acl.so res_pjsip_authenticator_digest.so res_pjsip_caller_id.so res_pjsip_config_wizard.so res_pjsip_dialog_info_body_generator.so res_pjsip_diversion.so res_pjsip_dlg_options.so res_pjsip_dtmf_info.so res_pjsip_endpoint_identifier_anonymous.so res_pjsip_endpoint_identifier_ip.so res_pjsip_endpoint_identifier_user.so res_pjsip_exten_state.so res_pjsip_header_funcs.so res_pjsip_keepalive.so res_pjsip_log_forwarder.so res_pjsip_logger.so res_pjsip_messaging.so res_pjsip_multihomed.so res_pjsip_mwi_body_generator.so res_pjsip_mwi.so res_pjsip_nat.so res_pjsip_notify.so res_pjsip_one_touch_record_info.so res_pjsip_outbound_authenticator_digest.so res_pjsip_outbound_publish.so res_pjsip_outbound_registration.so res_pjsip_path.so res_pjsip_phoneprov_provider.so res_pjsip_pidf_body_generator.so res_pjsip_pidf_digium_body_supplement.so res_pjsip_pidf_eyebeam_body_supplement.so res_pjsip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.so res_pjsip_send_to_voicemail.so res_pjsip_session.so res_pjsip_sips_contact.so res_pjsip.so res_pjsip_t38.so res_pjsip_transport_websocket.so res_pjsip_xpidf_body_generator.so On Tue, Mar 31, 2015 at 5:39 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 31, 2015 at 9:00 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Thank you mathew, The pjproject was detected on the installation process. When I run Asterisk I see that pjsip modules are running. The dependency checking for Asterisk assumes that the Asterisk modules are all installed in the 'default' location: def _find_asterisk_module(self, name): Determine if an Asterisk module exists if Dependency.ast.original_astmoddir == : return False module = %s/%s.so % (Dependency.ast.original_astmoddir, name) if os.path.exists(module): return True return False The fact that this is finding some of your Asterisk modules (app_echo) but not the PJSIP ones is a bit odd. What is the output of: ls /usr/lib/asterisk/modules | grep pjsip -- 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 -- _ -- 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
Re: [asterisk-dev] running pjsip testsuite
Got it !!! The testsuite was looking for these modules in /usr/lib64. I recompiled the asterisk with --libdir=/usr/lib64 and it works. Now the test is running I will start working on it now. Thank you. On Tue, Mar 31, 2015 at 6:09 PM, Yaron Nachum nachum.ya...@gmail.com wrote: The following is the output [root@stnrd5652 testsuite]# ls /usr/lib/asterisk/modules | grep pjsip chan_pjsip.so func_pjsip_aor.so func_pjsip_contact.so func_pjsip_endpoint.so res_hep_pjsip.so res_pjsip_acl.so res_pjsip_authenticator_digest.so res_pjsip_caller_id.so res_pjsip_config_wizard.so res_pjsip_dialog_info_body_generator.so res_pjsip_diversion.so res_pjsip_dlg_options.so res_pjsip_dtmf_info.so res_pjsip_endpoint_identifier_anonymous.so res_pjsip_endpoint_identifier_ip.so res_pjsip_endpoint_identifier_user.so res_pjsip_exten_state.so res_pjsip_header_funcs.so res_pjsip_keepalive.so res_pjsip_log_forwarder.so res_pjsip_logger.so res_pjsip_messaging.so res_pjsip_multihomed.so res_pjsip_mwi_body_generator.so res_pjsip_mwi.so res_pjsip_nat.so res_pjsip_notify.so res_pjsip_one_touch_record_info.so res_pjsip_outbound_authenticator_digest.so res_pjsip_outbound_publish.so res_pjsip_outbound_registration.so res_pjsip_path.so res_pjsip_phoneprov_provider.so res_pjsip_pidf_body_generator.so res_pjsip_pidf_digium_body_supplement.so res_pjsip_pidf_eyebeam_body_supplement.so res_pjsip_publish_asterisk.so res_pjsip_pubsub.so res_pjsip_refer.so res_pjsip_registrar_expire.so res_pjsip_registrar.so res_pjsip_rfc3326.so res_pjsip_sdp_rtp.so res_pjsip_send_to_voicemail.so res_pjsip_session.so res_pjsip_sips_contact.so res_pjsip.so res_pjsip_t38.so res_pjsip_transport_websocket.so res_pjsip_xpidf_body_generator.so On Tue, Mar 31, 2015 at 5:39 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 31, 2015 at 9:00 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Thank you mathew, The pjproject was detected on the installation process. When I run Asterisk I see that pjsip modules are running. The dependency checking for Asterisk assumes that the Asterisk modules are all installed in the 'default' location: def _find_asterisk_module(self, name): Determine if an Asterisk module exists if Dependency.ast.original_astmoddir == : return False module = %s/%s.so % (Dependency.ast.original_astmoddir, name) if os.path.exists(module): return True return False The fact that this is finding some of your Asterisk modules (app_echo) but not the PJSIP ones is a bit odd. What is the output of: ls /usr/lib/asterisk/modules | grep pjsip -- 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 -- _ -- 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
Re: [asterisk-dev] running pjsip testsuite
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. Richard -- _ -- 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
Re: [asterisk-dev] running pjsip testsuite
Thank you mathew, The pjproject was detected on the installation process. When I run Asterisk I see that pjsip modules are running. Any idea? Yaron On Tue, Mar 31, 2015 at 4:12 PM, Matthew Jordan mjor...@digium.com wrote: On Tue, Mar 31, 2015 at 8:04 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hi everyone, I am trying to add some tests for the PJSIP auto-dtmf support . Before I start I just wanted to run some of the existing tests in order to understand the process. Whenever I try to run a test from the pjsip tests I get - --- -- Dependency: res_pjsip - False. The following output was generated when I tried to run the rpid_immediate test: [root@stnrd5652 testsuite]# ./runtests.py -t tests/channels/pjsip/rpid_immediate/ Running tests for Asterisk SVN-trunk-r431522M ... Tests to run: 0, Maximum test inactivity time: -1 sec. -- Cannot run test 'tests/channels/pjsip/rpid_immediate' --- -- Minimum Version: 13.4.0 (True) --- -- Maximum Version: (True) --- -- Tags: ['pjsip'] --- -- Dependency: twisted - True --- -- Dependency: starpy - True --- -- Dependency: app_dial - True --- -- Dependency: app_echo - True --- -- Dependency: func_callerid - True --- -- Dependency: chan_pjsip - False --- -- Dependency: res_pjsip - False --- -- Dependency: res_pjsip_caller_id - False --- -- Dependency: res_pjsip_endpoint_identifier_user - False --- -- Dependency: res_pjsip_sdp_rtp - False --- -- Dependency: res_pjsip_session - False ?xml version=1.0 encoding=utf-8? testsuite errors=0 failures=0 name=AsteriskTestSuite tests=0 time=0.00/ Am I doing anything stupid? Congratulations on getting this far! Looks like you've got most of the dependencies worked out in the testsuite. The dependency checking for 'res_pjsip' is looking at what modules you have installed on the system. Double check that your Asterisk installation did detect pjproject, and that it built and installed the res_pjsip* modules. You can check this in menuselect. Matt -- 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 -- _ -- 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
Re: [asterisk-dev] running pjsip testsuite
On Tue, Mar 31, 2015 at 9:00 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Thank you mathew, The pjproject was detected on the installation process. When I run Asterisk I see that pjsip modules are running. The dependency checking for Asterisk assumes that the Asterisk modules are all installed in the 'default' location: def _find_asterisk_module(self, name): Determine if an Asterisk module exists if Dependency.ast.original_astmoddir == : return False module = %s/%s.so % (Dependency.ast.original_astmoddir, name) if os.path.exists(module): return True return False The fact that this is finding some of your Asterisk modules (app_echo) but not the PJSIP ones is a bit odd. What is the output of: ls /usr/lib/asterisk/modules | grep pjsip -- 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
Re: [asterisk-dev] running pjsip testsuite
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
