[asterisk-dev] Change in asterisk[master]: res_pjsip: Add external PJSIP resolver implementation using ...
Mark Michelson has posted comments on this change.
Change subject: res_pjsip: Add external PJSIP resolver implementation using
core DNS API.
..
Patch Set 1:
(3 comments)
https://gerrit.asterisk.org/#/c/75/1/main/dns_query_set.c
File main/dns_query_set.c:
Line 112: if (query_set-in_progress) {
: return -1;
: }
I suggest having an error message here.
I'd actually go so far as to consider putting an assertion in here because it
likely means there is some sort of programmer error occurring if we are
attempting to add queries to a set after starting the resolution.
https://gerrit.asterisk.org/#/c/75/1/main/dns_recurring.c
File main/dns_recurring.c:
Line 36: #include asterisk/vector.h
: #include asterisk/sched.h
: #include asterisk/strings.h
: #include asterisk/dns_core.h
: #include asterisk/dns_recurring.h
: #include asterisk/dns_query_set.h
On reviewboard, I asked about why these includes got added, and the the issue
was marked as resolved, but they're still included now. Are they needed?
https://gerrit.asterisk.org/#/c/75/1/res/res_pjsip/pjsip_resolver.c
File res/res_pjsip/pjsip_resolver.c:
Line 544: if ((type == PJSIP_TRANSPORT_UNSPECIFIED
sip_transport_is_available(PJSIP_TRANSPORT_UDP6)) ||
: sip_transport_is_available(type +
PJSIP_TRANSPORT_IPV6)) {
: res |= sip_resolve_add(resolve, host, ns_t_,
ns_c_in, (type == PJSIP_TRANSPORT_UNSPECIFIED ? PJSIP_TRANSPORT_UDP6 : type +
PJSIP_TRANSPORT_IPV6), target-addr.port);
: }
:
: if ((type == PJSIP_TRANSPORT_UNSPECIFIED
sip_transport_is_available(PJSIP_TRANSPORT_UDP)) ||
: sip_transport_is_available(type)) {
: res |= sip_resolve_add(resolve, host, ns_t_a, ns_c_in,
(type == PJSIP_TRANSPORT_UNSPECIFIED ? PJSIP_TRANSPORT_UDP : type),
target-addr.port);
: }
After giving RFC 3263 another look, I don't think these A and lookups
should be done here.
Section 4.2 says If no SRV records were found, the client performs an A or
record lookup of the domain name.
Since you can't know if the SRV lookup(s) provided any records yet, performing
this lookup is premature.
Now, I guess you could go ahead and perform this lookup early, but if the SRV
lookup(s) provide any results, you can't actually use these A/ records.
--
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Gerrit-MessageType: comment
Gerrit-Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp jc...@digium.com
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
Gerrit-HasComments: Yes
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[asterisk-dev] Change in asterisk[1.8]: .gitignore: Ignore tarballs (*.gz)
Mark Michelson has posted comments on this change. Change subject: .gitignore: Ignore tarballs (*.gz) .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/86 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[12]: .gitignore: Ignore tarballs (*.gz)
Mark Michelson has posted comments on this change. Change subject: .gitignore: Ignore tarballs (*.gz) .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/85 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[1.8]: .gitignore: Ignore tarballs (*.gz)
Mark Michelson has submitted this change and it was merged. Change subject: .gitignore: Ignore tarballs (*.gz) .. .gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz extension. This will cause git to ignore downloaded sound tarballs in the the sounds/ directory. Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 --- M .gitignore 1 file changed, 1 insertion(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve Jared K. Smith: Looks good to me, but someone else must approve Corey Farrell: Looks good to me, but someone else must approve diff --git a/.gitignore b/.gitignore index cf46873..33dd7cc 100644 --- a/.gitignore +++ b/.gitignore @@ -9,6 +9,7 @@ *~ *.[oadi] +*.gz *.ii *.oo *.eo -- To view, visit https://gerrit.asterisk.org/86 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in asterisk[1.8]: Add .gitignore and .gitreview files
Mark Michelson has posted comments on this change. Change subject: Add .gitignore and .gitreview files .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/82 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[1.8]: Add .gitignore and .gitreview files
Mark Michelson has submitted this change and it was merged. Change subject: Add .gitignore and .gitreview files .. Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Tested-by: George Joseph (cherry picked from commit b35e184d41c4e61e98b455d70321ba90118600a1) --- A .gitignore A .gitreview A addons/.gitignore A agi/.gitignore A build_tools/.gitignore A doc/.gitignore A include/asterisk/.gitignore A main/.gitignore A menuselect/.gitignore A res/ael/.gitignore A utils/.gitignore 11 files changed, 72 insertions(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve diff --git a/.gitignore b/.gitignore new file mode 100644 index 000..cf46873 --- /dev/null +++ b/.gitignore @@ -0,0 +1,26 @@ +# git ls-files --others --exclude-from=.git/info/exclude +# Lines that start with '#' are comments. +# For a project mostly in C, the following would be a good set of +# exclude patterns (uncomment them if you want to use them): +# *.[oa] +# *~ + +# See .gitignore in subdirectories for more ignored files + +*~ +*.[oadi] +*.ii +*.oo +*.eo +*.so +*.exports +*.moduleinfo +*.makeopts +*.makedeps +makeopts +.lastclean +config.log +config.status +defaults.h +makeopts.embed_rules +menuselect-tree diff --git a/.gitreview b/.gitreview new file mode 100644 index 000..f9ef050 --- /dev/null +++ b/.gitreview @@ -0,0 +1,4 @@ +[gerrit] +host=gerrit.asterisk.org +port=29418 +project=asterisk.git diff --git a/addons/.gitignore b/addons/.gitignore new file mode 100644 index 000..663e668 --- /dev/null +++ b/addons/.gitignore @@ -0,0 +1 @@ +mp3 diff --git a/agi/.gitignore b/agi/.gitignore new file mode 100644 index 000..9b2a4e2 --- /dev/null +++ b/agi/.gitignore @@ -0,0 +1,3 @@ +eagi-sphinx-test +eagi-test +strcompat.c diff --git a/build_tools/.gitignore b/build_tools/.gitignore new file mode 100644 index 000..c60a0df --- /dev/null +++ b/build_tools/.gitignore @@ -0,0 +1 @@ +menuselect-deps diff --git a/doc/.gitignore b/doc/.gitignore new file mode 100644 index 000..27acdb3 --- /dev/null +++ b/doc/.gitignore @@ -0,0 +1 @@ +core-en_US.xml diff --git a/include/asterisk/.gitignore b/include/asterisk/.gitignore new file mode 100644 index 000..ae33b3c --- /dev/null +++ b/include/asterisk/.gitignore @@ -0,0 +1,3 @@ +autoconfig.h +build.h +buildopts.h diff --git a/main/.gitignore b/main/.gitignore new file mode 100644 index 000..23f5c58 --- /dev/null +++ b/main/.gitignore @@ -0,0 +1,3 @@ +asterisk +libasteriskssl.so.1 +version.c diff --git a/menuselect/.gitignore b/menuselect/.gitignore new file mode 100644 index 000..38ea2d3 --- /dev/null +++ b/menuselect/.gitignore @@ -0,0 +1,5 @@ +autoconfig.h +cmenuselect +config.log +config.status +menuselect diff --git a/res/ael/.gitignore b/res/ael/.gitignore new file mode 100644 index 000..f39b612 --- /dev/null +++ b/res/ael/.gitignore @@ -0,0 +1 @@ +ael.output diff --git a/utils/.gitignore b/utils/.gitignore new file mode 100644 index 000..ed37a06 --- /dev/null +++ b/utils/.gitignore @@ -0,0 +1,24 @@ +aelbison.c +aelparse +aelparse.c +ast_expr2.c +ast_expr2f.c +astman +astcanary +astdb2bdb +astdb2sqlite3 +check_expr +check_expr2 +conf2ael +db1-ast/libdb1.a +hashtab.c +lock.c +md5.c +muted +pbx_ael.c +pval.c +smsq +stereorize +strcompat.c +streamplayer +threadstorage.c -- To view, visit https://gerrit.asterisk.org/82 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in asterisk[1.8]: main/editline: Add .gitignore.
Mark Michelson has submitted this change and it was merged. Change subject: main/editline: Add .gitignore. .. main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d (cherry picked from commit 5d34bce635ad7f225d557fa226e6b79c6293dbe1) --- A main/editline/.gitignore 1 file changed, 13 insertions(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve diff --git a/main/editline/.gitignore b/main/editline/.gitignore new file mode 100644 index 000..d3bb06b --- /dev/null +++ b/main/editline/.gitignore @@ -0,0 +1,13 @@ +*.o_a +Makefile +common.h +config.cache +config.h +editline.c +emacs.h +fcns.c +fcns.h +help.c +help.h +makelist +vi.h -- To view, visit https://gerrit.asterisk.org/84 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in asterisk[1.8]: main/editline: Add .gitignore.
Mark Michelson has posted comments on this change. Change subject: main/editline: Add .gitignore. .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/84 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 1.8 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[12]: main/editline: Add .gitignore.
Mark Michelson has posted comments on this change. Change subject: main/editline: Add .gitignore. .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/83 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[12]: main/editline: Add .gitignore.
Mark Michelson has submitted this change and it was merged. Change subject: main/editline: Add .gitignore. .. main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d (cherry picked from commit 5d34bce635ad7f225d557fa226e6b79c6293dbe1) --- A main/editline/.gitignore 1 file changed, 13 insertions(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve diff --git a/main/editline/.gitignore b/main/editline/.gitignore new file mode 100644 index 000..d3bb06b --- /dev/null +++ b/main/editline/.gitignore @@ -0,0 +1,13 @@ +*.o_a +Makefile +common.h +config.cache +config.h +editline.c +emacs.h +fcns.c +fcns.h +help.c +help.h +makelist +vi.h -- To view, visit https://gerrit.asterisk.org/83 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in asterisk[12]: Add .gitignore and .gitreview files
Mark Michelson has submitted this change and it was merged. Change subject: Add .gitignore and .gitreview files .. Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Tested-by: George Joseph (cherry picked from commit b35e184d41c4e61e98b455d70321ba90118600a1) --- A .gitignore A .gitreview A addons/.gitignore A agi/.gitignore A build_tools/.gitignore A doc/.gitignore A include/asterisk/.gitignore A main/.gitignore A menuselect/.gitignore A res/ael/.gitignore A utils/.gitignore 11 files changed, 72 insertions(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve diff --git a/.gitignore b/.gitignore new file mode 100644 index 000..cf46873 --- /dev/null +++ b/.gitignore @@ -0,0 +1,26 @@ +# git ls-files --others --exclude-from=.git/info/exclude +# Lines that start with '#' are comments. +# For a project mostly in C, the following would be a good set of +# exclude patterns (uncomment them if you want to use them): +# *.[oa] +# *~ + +# See .gitignore in subdirectories for more ignored files + +*~ +*.[oadi] +*.ii +*.oo +*.eo +*.so +*.exports +*.moduleinfo +*.makeopts +*.makedeps +makeopts +.lastclean +config.log +config.status +defaults.h +makeopts.embed_rules +menuselect-tree diff --git a/.gitreview b/.gitreview new file mode 100644 index 000..f9ef050 --- /dev/null +++ b/.gitreview @@ -0,0 +1,4 @@ +[gerrit] +host=gerrit.asterisk.org +port=29418 +project=asterisk.git diff --git a/addons/.gitignore b/addons/.gitignore new file mode 100644 index 000..663e668 --- /dev/null +++ b/addons/.gitignore @@ -0,0 +1 @@ +mp3 diff --git a/agi/.gitignore b/agi/.gitignore new file mode 100644 index 000..9b2a4e2 --- /dev/null +++ b/agi/.gitignore @@ -0,0 +1,3 @@ +eagi-sphinx-test +eagi-test +strcompat.c diff --git a/build_tools/.gitignore b/build_tools/.gitignore new file mode 100644 index 000..c60a0df --- /dev/null +++ b/build_tools/.gitignore @@ -0,0 +1 @@ +menuselect-deps diff --git a/doc/.gitignore b/doc/.gitignore new file mode 100644 index 000..27acdb3 --- /dev/null +++ b/doc/.gitignore @@ -0,0 +1 @@ +core-en_US.xml diff --git a/include/asterisk/.gitignore b/include/asterisk/.gitignore new file mode 100644 index 000..ae33b3c --- /dev/null +++ b/include/asterisk/.gitignore @@ -0,0 +1,3 @@ +autoconfig.h +build.h +buildopts.h diff --git a/main/.gitignore b/main/.gitignore new file mode 100644 index 000..23f5c58 --- /dev/null +++ b/main/.gitignore @@ -0,0 +1,3 @@ +asterisk +libasteriskssl.so.1 +version.c diff --git a/menuselect/.gitignore b/menuselect/.gitignore new file mode 100644 index 000..38ea2d3 --- /dev/null +++ b/menuselect/.gitignore @@ -0,0 +1,5 @@ +autoconfig.h +cmenuselect +config.log +config.status +menuselect diff --git a/res/ael/.gitignore b/res/ael/.gitignore new file mode 100644 index 000..f39b612 --- /dev/null +++ b/res/ael/.gitignore @@ -0,0 +1 @@ +ael.output diff --git a/utils/.gitignore b/utils/.gitignore new file mode 100644 index 000..ed37a06 --- /dev/null +++ b/utils/.gitignore @@ -0,0 +1,24 @@ +aelbison.c +aelparse +aelparse.c +ast_expr2.c +ast_expr2f.c +astman +astcanary +astdb2bdb +astdb2sqlite3 +check_expr +check_expr2 +conf2ael +db1-ast/libdb1.a +hashtab.c +lock.c +md5.c +muted +pbx_ael.c +pval.c +smsq +stereorize +strcompat.c +streamplayer +threadstorage.c -- To view, visit https://gerrit.asterisk.org/81 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in asterisk[12]: Add .gitignore and .gitreview files
Mark Michelson has posted comments on this change. Change subject: Add .gitignore and .gitreview files .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/81 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I842a1588ff27d8a0189f12d597f0a7af033d6c69 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in repotools[master]: mapmantis: Remove dependency on digium_jira
Mark Michelson has posted comments on this change. Change subject: mapmantis: Remove dependency on digium_jira .. Patch Set 2: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/68 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0ae337d24e3db7d552406a57a740d260d5c4d2d7 Gerrit-PatchSet: 2 Gerrit-Project: repotools Gerrit-Branch: master Gerrit-Owner: Matt Jordan mjor...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in repotools[master]: digium_jira: Refactor module to wrap the Atlassian JIRA REST...
Hello Jared K. Smith, I'd like you to reexamine a change. Please visit https://gerrit.asterisk.org/69 to look at the new patch set (#2). Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST client .. digium_jira: Refactor module to wrap the Atlassian JIRA REST client Many of our scripts that use the JIRA REST Python client first have to obtain credentials from the .jira_login file, or else prompt for them. The act of getting credentials and setting up the basic client is done in a number of scripts. Rather than reproduce that logic across many modules, this patch takes the defunct digium_jira module and repurposes it to contain those functions. Since Atlassian no longer supports its SOAP API and we no longer use it in any of our modules, its reasonable to locate this functionality in this module. Change-Id: I69932dd472aef4290af97e809ce6b9ec9c25b39d --- M digium_jira.py 1 file changed, 30 insertions(+), 173 deletions(-) git pull ssh://gerrit.asterisk.org:29418/repotools refs/changes/69/69/2 -- To view, visit https://gerrit.asterisk.org/69 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I69932dd472aef4290af97e809ce6b9ec9c25b39d Gerrit-PatchSet: 2 Gerrit-Project: repotools Gerrit-Branch: master Gerrit-Owner: Matt Jordan mjor...@digium.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net -- _ -- 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
[asterisk-dev] Change in asterisk[master]: Fixing extconf compile
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing utils to fail to compile.
Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
---
M utils/extconf.c
1 file changed, 1 insertion(+), 1 deletion(-)
Approvals:
Mark Michelson: Looks good to me, approved; Verified
Jared K. Smith: Looks good to me, but someone else must approve
diff --git a/utils/extconf.c b/utils/extconf.c
index 7ddbb67..53347b4 100644
--- a/utils/extconf.c
+++ b/utils/extconf.c
@@ -503,7 +503,7 @@
static void __attribute__((constructor)) init_##mutex(void) \
{ \
ast_mutex_init(mutex); \
-} \
+}
#else /* !AST_MUTEX_INIT_W_CONSTRUCTORS */
/* By default, use static initialization of mutexes. */
#define __AST_MUTEX_DEFINE(scope, mutex) \
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: David M. Lee d...@digium.com
Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
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[asterisk-dev] Change in asterisk[13]: Fixing extconf compile
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing utils to fail to compile.
Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
---
M utils/extconf.c
1 file changed, 1 insertion(+), 1 deletion(-)
Approvals:
Mark Michelson: Looks good to me, approved; Verified
Jared K. Smith: Looks good to me, but someone else must approve
diff --git a/utils/extconf.c b/utils/extconf.c
index 89cf6dd..7989bcd 100644
--- a/utils/extconf.c
+++ b/utils/extconf.c
@@ -502,7 +502,7 @@
static void __attribute__((constructor)) init_##mutex(void) \
{ \
ast_mutex_init(mutex); \
-} \
+}
#else /* !AST_MUTEX_INIT_W_CONSTRUCTORS */
/* By default, use static initialization of mutexes. */
#define __AST_MUTEX_DEFINE(scope, mutex) \
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: David M. Lee d...@digium.com
Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
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[asterisk-dev] Change in asterisk[11]: Fixing extconf compile
Mark Michelson has submitted this change and it was merged.
Change subject: Fixing extconf compile
..
Fixing extconf compile
During the mass code deletion for clang support, a stray backslash was
left behind that was causing utils to fail to compile.
Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
---
M utils/extconf.c
1 file changed, 1 insertion(+), 1 deletion(-)
Approvals:
Mark Michelson: Looks good to me, approved; Verified
Jared K. Smith: Looks good to me, but someone else must approve
diff --git a/utils/extconf.c b/utils/extconf.c
index 2b1bc0c..bd2cc9a 100644
--- a/utils/extconf.c
+++ b/utils/extconf.c
@@ -487,7 +487,7 @@
static void __attribute__((constructor)) init_##mutex(void) \
{ \
ast_mutex_init(mutex); \
-} \
+}
#else /* !AST_MUTEX_INIT_W_CONSTRUCTORS */
/* By default, use static initialization of mutexes. */
#define __AST_MUTEX_DEFINE(scope, mutex) \
--
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Gerrit-MessageType: merged
Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 11
Gerrit-Owner: David M. Lee d...@digium.com
Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
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[asterisk-dev] Change in asterisk[master]: Fixing extconf compile
Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/90 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: master Gerrit-Owner: David M. Lee d...@digium.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[13]: Fixing extconf compile
Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/89 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 13 Gerrit-Owner: David M. Lee d...@digium.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[11]: Fixing extconf compile
Mark Michelson has posted comments on this change. Change subject: Fixing extconf compile .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/88 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 11 Gerrit-Owner: David M. Lee d...@digium.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in asterisk[12]: .gitignore: Ignore tarballs (*.gz)
Mark Michelson has submitted this change and it was merged. Change subject: .gitignore: Ignore tarballs (*.gz) .. .gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz extension. This will cause git to ignore downloaded sound tarballs in the the sounds/ directory. Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 --- M .gitignore 1 file changed, 1 insertion(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Matt Jordan: Looks good to me, but someone else must approve Jared K. Smith: Looks good to me, but someone else must approve diff --git a/.gitignore b/.gitignore index cf46873..33dd7cc 100644 --- a/.gitignore +++ b/.gitignore @@ -9,6 +9,7 @@ *~ *.[oadi] +*.gz *.ii *.oo *.eo -- To view, visit https://gerrit.asterisk.org/85 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: Ie84f085cc0fa51262209e7bfc1b1ba8c04a1ef59 Gerrit-PatchSet: 1 Gerrit-Project: asterisk Gerrit-Branch: 12 Gerrit-Owner: George Joseph george.jos...@fairview5.com Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in repotools[master]: digium_jira: Refactor module to wrap the Atlassian JIRA REST...
Mark Michelson has posted comments on this change.
Change subject: digium_jira: Refactor module to wrap the Atlassian JIRA REST
client
..
Patch Set 2:
(2 comments)
https://gerrit.asterisk.org/#/c/69/2/digium_jira.py
File digium_jira.py:
Line 16: try:
: jira_cache = open(os.path.expanduser('~') + /.jira_login,
r)
: jira_user = jira_cache.readline().strip()
: jira_pw = jira_cache.readline().strip()
: jira_cache.close()
: return (jira_user, jira_pw)
: except IOError:
: pass
Python developers are encouraged to use the with keyword for file I/O since
it automatically will close the file properly even if an exception occurs while
operating on the file.
Line 25:# Didn't get auth deatils from file, try interactive instead.
s/deatils/details/
--
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Gerrit-Change-Id: I69932dd472aef4290af97e809ce6b9ec9c25b39d
Gerrit-PatchSet: 2
Gerrit-Project: repotools
Gerrit-Branch: master
Gerrit-Owner: Matt Jordan mjor...@digium.com
Gerrit-Reviewer: Jared K. Smith jaredsm...@jaredsmith.net
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
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[asterisk-dev] Change in testsuite[master]: rest_api/applications/stasisstatus: Make run-test executable
Mark Michelson has posted comments on this change. Change subject: rest_api/applications/stasisstatus: Make run-test executable .. Patch Set 1: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/36 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: Ibf0647d37c9bba0318f251e1040c03372bce1b54 Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Matt Jordan mjor...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: rest_api/applications/stasisstatus: Make run-test executable
Mark Michelson has submitted this change and it was merged. Change subject: rest_api/applications/stasisstatus: Make run-test executable .. rest_api/applications/stasisstatus: Make run-test executable If it isn't executable, it can't run! Change-Id: Ibf0647d37c9bba0318f251e1040c03372bce1b54 --- M tests/rest_api/applications/stasisstatus/run-test 1 file changed, 0 insertions(+), 0 deletions(-) Approvals: Mark Michelson: Looks good to me, approved; Verified Ashley Sanders: Looks good to me, but someone else must approve diff --git a/tests/rest_api/applications/stasisstatus/run-test b/tests/rest_api/applications/stasisstatus/run-test old mode 100644 new mode 100755 -- To view, visit https://gerrit.asterisk.org/36 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: merged Gerrit-Change-Id: Ibf0647d37c9bba0318f251e1040c03372bce1b54 Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Matt Jordan mjor...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Verified+1 -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 5 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Code-Review+2 -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 5 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: -Verified Removing the Verified since there are still outstanding findings. -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 5 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 5: Code-Review+1 (2 comments) It's looking good by me! https://gerrit.asterisk.org/#/c/18/5/tests/rest_api/applications/stasisstatus/ari_client.py File tests/rest_api/applications/stasisstatus/ari_client.py: Line 365:if run is 0: : LOGGER.debug(msg + 'Tearing down active connections.') : self.__delete_all_channels() : reactor.callLater(2, wait_for_it, deferred, 1) : elif run is 1: : if len(self.__channels) 0: : msg += 'Waiting for channels to be destroyed.' : LOGGER.debug(msg) : reactor.callLater(2, wait_for_it, deferred, 1) : reactor.callLater(2, wait_for_it, deferred, 2) : elif run is 2: : LOGGER.debug(msg + 'Disconnecting web socket.') : self.__ari = None : self.__factory = None : self.disconnect_websocket() : reactor.callLater(2, wait_for_it, deferred, 3) : elif run is 3: This is another of those cases where I'm not 100% sure on this, but I think the use of the 'is' operator in these comparisons is iffy. The 'is' operator returns true if the operands on either side refer to the same object, not if their values are equal. I'm honestly not sure what the behavior is when an integer literal is used as an operand, though. If I were writing this, though, I'd default to the safe bet and use the '==' operator for these comparisons instead. https://gerrit.asterisk.org/#/c/18/5/tests/rest_api/applications/stasisstatus/configs/ast1/sip.conf File tests/rest_api/applications/stasisstatus/configs/ast1/sip.conf: Line 9: [acme](!) I am not sure that this template or the sherman endpoint is being used any In fact, I don't think SIP is being used at all for your tests, so you should be able to just scrap this conf file entirely. -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 5 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: Yes -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 10: Code-Review+2 Verified+1 -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 10 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has submitted this change and it was merged.
Change subject: stasis: set a channel variable on websocket disconnect error
..
stasis: set a channel variable on websocket disconnect error
This test is to ensure Asterisk applies the correct state to the channel
variable, STASISSTATUS. STASISSTATUS was introduced as a means for Stasis
applications to have context when errors occur in Stasis that disrupt normal
processing.
The test scenarios:
1. The 'Babs' scenario:
a. A channel is originated through ARI referencing a subscribed app (Babs)
that was registered in Stasis during startup.
b. After Stasis is started, the channel is then hungup.
c. A check is made to ensure that the value of STASISSTATUS is SUCCESS.
2. The 'Bugs' scenario:
a. A channel is originated through ARI referencing a subscribed app
(BugsAlt) that was never registered in Stasis.
b. A check is then made to ensure that the value of STASISSTATUS is FAILED.
3. The 'Buster' scenario:
a. A channel is originated through ARI referencing a subscribed app
(Buster) that was registered in Stasis during startup.
b. While the channel from step 'a' is still active, the websocket is then
disconnected out from underneath ARI.
c. A new channel is originated through ARI, also referencing the subscribed
app (Buster) that was registered in Stasis during startup.
d. A check is then made to ensure that the value of STASISSTATUS is FAILED.
ASTERISK-24802
Reported By: Kevin Harwell
Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13
---
A tests/rest_api/applications/stasisstatus/__init__.py
A tests/rest_api/applications/stasisstatus/ari_client.py
A tests/rest_api/applications/stasisstatus/configs/ast1/extensions.conf
A tests/rest_api/applications/stasisstatus/monitor.py
A tests/rest_api/applications/stasisstatus/observable_object.py
A tests/rest_api/applications/stasisstatus/run-test
A tests/rest_api/applications/stasisstatus/test-config.yaml
A tests/rest_api/applications/stasisstatus/test_case.py
A tests/rest_api/applications/stasisstatus/test_scenario.py
M tests/rest_api/applications/tests.yaml
10 files changed, 1,447 insertions(+), 0 deletions(-)
Approvals:
Mark Michelson: Looks good to me, approved; Verified
Matt Jordan: Looks good to me, but someone else must approve
diff --git a/tests/rest_api/applications/stasisstatus/__init__.py
b/tests/rest_api/applications/stasisstatus/__init__.py
new file mode 100644
index 000..e69de29
--- /dev/null
+++ b/tests/rest_api/applications/stasisstatus/__init__.py
diff --git a/tests/rest_api/applications/stasisstatus/ari_client.py
b/tests/rest_api/applications/stasisstatus/ari_client.py
new file mode 100644
index 000..e7e2eb3
--- /dev/null
+++ b/tests/rest_api/applications/stasisstatus/ari_client.py
@@ -0,0 +1,404 @@
+#!/usr/bin/env python
+
+Copyright (C) 2015, Digium, Inc.
+Ashley Sanders asand...@digium.com
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+
+
+import sys
+import logging
+import uuid
+
+sys.path.append(lib/python)
+sys.path.append(tests/rest_api/applications)
+
+from asterisk.ari import ARI, AriClientFactory
+from stasisstatus.observable_object import ObservableObject
+from twisted.internet import defer, reactor
+
+LOGGER = logging.getLogger(__name__)
+
+
+class AriClient(ObservableObject):
+The ARI client.
+
+ This class serves as a facade for ARI and AriClientFactory. It is
+ responsible for creating and persisting the connection state needed to
+ execute a test scenario.
+
+
+def __init__(self, host, port, credentials, name='testsuite'):
+Constructor.
+
+Keyword Arguments:
+host -- The [bindaddr] of the Asterisk HTTP web
+ server.
+port -- The [bindport] of the Asterisk HTTP web
+ server.
+credentials -- User credentials for ARI. A tuple.
+ E.g.: ('username', 'password').
+name -- The name of the app to register in Stasis via
+ ARI (optional) (default 'testsuite').
+
+
+
+super(AriClient, self).__init__(name, ['on_channelcreated',
+ 'on_channeldestroyed',
+ 'on_channelvarset',
+ 'on_client_start',
+ 'on_client_stop',
+ 'on_stasisend',
+ 'on_stasisstart',
+ 'on_ws_open',
+ 'on_ws_closed'])
+self.__ari = None
+self.__factory = None
+
[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.
Mark Michelson has uploaded a new patch set (#3). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test was recently rewritten to prevent it from bouncing during automatic test runs. The rewrite attempted to break into two tests in an attempt to separate the logic of different versions from one another. Ashley pointed out on my original Asterisk 11 version of the patch that there are only small difference between the Asterisk 11 and 12 versions of the test, resulting in a lot of repeated boilerplate code that could otherwise be avoided. This change alters the 12+ specific test by separating the bridge logic for different versions into their own classes. I have verified that the test passes using both Asterisk 11 and Asterisk 13. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Fix a typo where a function call was evaluated as a boolean. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Address review feedback from Ashley and Matt * Make efforts to make pylint less angry * Add debugging to the 11 version since it's weird * Changed some variable names. I went with the suggested names in some case, but in others I did not. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d --- M tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/test-config.yaml 2 files changed, 199 insertions(+), 45 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/29/29/3 -- To view, visit https://gerrit.asterisk.org/29 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.
Mark Michelson has uploaded a new patch set (#3). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test was recently rewritten to prevent it from bouncing during automatic test runs. The rewrite attempted to break into two tests in an attempt to separate the logic of different versions from one another. Ashley pointed out on my original Asterisk 11 version of the patch that there are only small difference between the Asterisk 11 and 12 versions of the test, resulting in a lot of repeated boilerplate code that could otherwise be avoided. This change alters the 12+ specific test by separating the bridge logic for different versions into their own classes. I have verified that the test passes using both Asterisk 11 and Asterisk 13. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Fix a typo where a function call was evaluated as a boolean. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Address review feedback from Ashley and Matt * Make efforts to make pylint less angry * Add debugging to the 11 version since it's weird * Changed some variable names. I went with the suggested names in some case, but in others I did not. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d --- M tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/test-config.yaml 2 files changed, 199 insertions(+), 45 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/29/29/3 -- To view, visit https://gerrit.asterisk.org/29 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Enable support for directory containing custom tests.
Mark Michelson has posted comments on this change. Change subject: Enable support for directory containing custom tests. .. Patch Set 1: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/27 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: Iff45fe5574900b5c5c77e3132984659133baadfd Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Corey Farrell g...@cfware.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Testsuite: New test for FAX via PJSIP T38 with authentication
Mark Michelson has posted comments on this change. Change subject: Testsuite: New test for FAX via PJSIP T38 with authentication .. Patch Set 4: Code-Review+2 -- To view, visit https://gerrit.asterisk.org/28 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: If37cf20857ae3c0b35e0637a0a2cb7e7d6226df6 Gerrit-PatchSet: 4 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Jonathan Rose jr...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Jonathan Rose jr...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Add SIP attended transfer for Asterisk 11.
Mark Michelson has abandoned this change. Change subject: Add SIP attended transfer for Asterisk 11. .. Abandoned -- To view, visit https://gerrit.asterisk.org/20 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: abandon Gerrit-Change-Id: I48c7b6a9298552aa756d0c2f26afbd6a96d553b5 Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Add SIP attended transfer for Asterisk 11.
Mark Michelson has posted comments on this change. Change subject: Add SIP attended transfer for Asterisk 11. .. Patch Set 1: I am abandoning this change in favor of change /c/29/ -- To view, visit https://gerrit.asterisk.org/20 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I48c7b6a9298552aa756d0c2f26afbd6a96d553b5 Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Corey Farrell g...@cfware.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.
Mark Michelson has posted comments on this change. Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. Patch Set 1: (1 comment) https://gerrit.asterisk.org/#/c/29/1/tests/channels/SIP/sip_attended_transfer/attended_transfer.py File tests/channels/SIP/sip_attended_transfer/attended_transfer.py: Line 235: if self.bridge_state.bridge2_bridged and self.carol_call_answered: Got a typo here. This should be self.bridge_staet.bridge2_bridged() (note the added parentheses) As it turns out my test runs must have been passing because it just so happened that bridge 2 was actually bridged before Alice's call to Carol had been answered. -- To view, visit https://gerrit.asterisk.org/29 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: Yes -- _ -- 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
[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.
Mark Michelson has uploaded a new change for review.
https://gerrit.asterisk.org/29
Change subject: sip_attended_transfer now supports pre-12 Asterisk versions.
..
sip_attended_transfer now supports pre-12 Asterisk versions.
The sip_attended transfer test was recently rewritten to prevent it from
bouncing during automatic test runs. The rewrite attempted to break into
two tests in an attempt to separate the logic of different versions from
one another.
Ashley pointed out on my original Asterisk 11 version of the patch that
there are only small difference between the Asterisk 11 and 12 versions
of the test, resulting in a lot of repeated boilerplate code that could
otherwise be avoided.
This change alters the 12+ specific test by separating the bridge logic
for different versions into their own classes. I have verified that the
test passes using both Asterisk 11 and Asterisk 13.
Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d
---
M tests/channels/SIP/sip_attended_transfer/attended_transfer.py
M tests/channels/SIP/sip_attended_transfer/test-config.yaml
2 files changed, 125 insertions(+), 45 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/29/29/1
diff --git a/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
b/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
index cb133f3..ef9f314 100644
--- a/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
+++ b/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
@@ -8,7 +8,11 @@
import logging
import pjsua as pj
+import sys
+sys.path.append('lib/python/asterisk')
+
+from version import AsteriskVersion
from twisted.internet import reactor
LOGGER = logging.getLogger(__name__)
@@ -52,8 +56,8 @@
reactor.callFromThread(self.on_answered)
-class BridgeState(object):
-'''Object for tracking state of a bridge
+class BridgeTwelve(object):
+'''Object for tracking attributes of an Asterisk 12+ bridge
The main data the test cares about is the bridge's unique id and whether
two
channels have been bridged together by the bridge.
@@ -61,6 +65,118 @@
def __init__(self):
self.unique_id = None
self.bridged = False
+
+
+class BridgeStateTwelve(object):
+'''Tracker of Bridge State for Asterisk 12+
+
+Since Asterisk 12+ has the concept of Bridge objects, this tracks the
+bridges by detecting when they are created. Once bridges are created, we
+determine that channels are bridged when BridgeEnter events indicate that
+two channels are present.
+'''
+def __init__(self, test_object, controller, ami):
+self.test_object = test_object
+self.controller = controller
+self.ami = ami
+self.bridge1 = BridgeTwelve()
+self.bridge2 = BridgeTwelve()
+
+self.ami.registerEvent('BridgeCreate', self.bridge_create)
+self.ami.registerEvent('BridgeEnter', self.bridge_enter)
+
+def bridge_create(self, ami, event):
+if not self.bridge1.unique_id:
+self.bridge1.unique_id = event.get('bridgeuniqueid')
+elif not self.bridge2.unique_id:
+self.bridge2.unique_id = event.get('bridgeuniqueid')
+else:
+LOGGER.error(Unexpected third bridge created)
+self.test_object.set_passed(False)
+self.test_object.stop_reactor()
+
+def bridge_enter(self, ami, event):
+if (event.get('bridgeuniqueid') == self.bridge1.unique_id and
+event.get('bridgenumchannels') == '2'):
+self.bridge1.bridged = True
+if self.controller.state == BOB_CALLED:
+self.controller.call_carol()
+elif self.controller.state == TRANSFERRED:
+self.controller.hangup_calls()
+else:
+LOGGER.error(Unexpected BridgeEnter event)
+self.test_object.set_passed(False)
+self.test_object.stop_reactor()
+elif (event.get('bridgeuniqueid') == self.bridge2.unique_id and
+ event.get('bridgenumchannels') == '2'):
+self.bridge2.bridged = True
+if self.controller.state == CAROL_CALLED:
+self.controller.transfer_call()
+elif self.controller.state == TRANSFERRED:
+self.controller.hangup_calls()
+else:
+LOGGER.error(Unexpected BridgeEnter event)
+self.test_object.set_passed(False)
+self.test_object.stop_reactor()
+
+def bridge1_bridged(self):
+return self.bridge1.bridged
+
+def bridge2_bridged(self):
+return self.bridge2.bridged
+
+
+class BridgeStateEleven(object):
+'''Tracker of bridge state for Asterisk 11-
+
+Since in Asterisk versions prior to 12, there are no bridge objects, the
+only way we can track the state of bridges in Asterisk is via Bridge
[asterisk-dev] Change in testsuite[master]: sip_attended_transfer now supports pre-12 Asterisk versions.
Mark Michelson has uploaded a new patch set (#2). Change subject: sip_attended_transfer now supports pre-12 Asterisk versions. .. sip_attended_transfer now supports pre-12 Asterisk versions. The sip_attended transfer test was recently rewritten to prevent it from bouncing during automatic test runs. The rewrite attempted to break into two tests in an attempt to separate the logic of different versions from one another. Ashley pointed out on my original Asterisk 11 version of the patch that there are only small difference between the Asterisk 11 and 12 versions of the test, resulting in a lot of repeated boilerplate code that could otherwise be avoided. This change alters the 12+ specific test by separating the bridge logic for different versions into their own classes. I have verified that the test passes using both Asterisk 11 and Asterisk 13. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Fix a typo where a function call was evaluated as a boolean. Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d --- M tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/test-config.yaml 2 files changed, 125 insertions(+), 45 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/29/29/2 -- To view, visit https://gerrit.asterisk.org/29 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I958c52cebb94f9cfc8dc8ed81311ae62efb2679d Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change. Change subject: stasis: set a channel variable on websocket disconnect error .. Patch Set 1: Code-Review+1 (4 comments) I noticed that I gave the Code-Review a 0 last time. I figure that a +1 is actually in order since I noticed that my comments last time weren't about functional deficiencies so much as style and nitpicks. So here's my +1. https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/ari_client.py File tests/rest_api/applications/stasisstatus/ari_client.py: Line 148: def on_channelcreated(self, message): This function is what builds up the local container for channels created by Ah, I missed the getattr() call in on_ws_event(). I'm going to blame gerrit's hijacking of my browser's search :-P https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/observable_object.py File tests/rest_api/applications/stasisstatus/observable_object.py: Line 84: del self.__registrar[event][:] From the python documentation https://docs.python.org/3/tutorial/datastruct Ah, thanks for the enlightenment! Line 132: if self.__suspended 0: : self.__suspended = 0 Not including the = sign excludes this redundant case: Correct, but my point was that if self.__suspended is 0 when entering this function, the result will be that it ends up being set to -1. My assumption was that you wanted self.__suspended to have a lower bound of 0. My suggestion was made in the spirit of brevity, but you could just as easily have if self.__suspended == 0: return elif self.__suspended 0: self.__suspended -= 1 else: # WTF, how did it get to be 0? Line 143: def __validate(self, **kwargs): It is so that there are more meaningful messages in the logfile when debugg I wasn't suggesting getting rid of the method, just rewriting its body. While DRY is fantastic, I'm also a firm believer in YAGNI, especially for tests that are concentrated on a specific case, where the data that is to be used is self-generated and known in advance. In this case, you've written a generic validation function that can take any number of named parameters, and based on the name of the parameter, perform some specific action. In actual use, the only parameter ever passed to this function is a single parameter called event (never multiple parameters), and it will never be NoneType (since the data being passed to this function is generated by your own test cases), and I don't actually think that any of the operations you perform could throw an IndexError anway. There's no reason to build a trans-oceanic cruise liner to cross a creek. The error message argument makes sense. If you want to keep an error message in here, that's cool. -- To view, visit https://gerrit.asterisk.org/18 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13 Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: Yes -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.
Hello Ashley Sanders, I'd like you to reexamine a change. Please visit https://gerrit.asterisk.org/19 to look at the new patch set (#3). Change subject: Rewrite sip_attended_transfer test to stop failing. .. Rewrite sip_attended_transfer test to stop failing. The sip_attended_transfer test has been bouncing for a while. There are two major fixes introduced here to prevent the bouncing. First, by converting to using the testsuite's PJSUA module, we no longer are using the native Python threading library. Instead, we are using a method that works better with the Twisted framework. Second, the test is more strict about when the transfer may be performed. The previous test would attempt the transfer when Asterisk reported that the call was bridged. The problem is that Asterisk may report the call as bridged before PJSUA has properly processed the 200 OK that Asterisk has sent to it. By waiting, we can be sure that all parties are prepared when the transfer is attempted. The test has also been rewritten to only work with Asterisk 12+. A new separate test will be written to work on Asterisk 11. This helps the code to be a little less cluttered. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Address review feedback from Ashley * Two CallCallback classes have been combined into one * Bridge state has been factored into a minimal class * Unnecessary checks of test state have been removed. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e --- A tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/configs/ast1/extensions.conf M tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf D tests/channels/SIP/sip_attended_transfer/run-test M tests/channels/SIP/sip_attended_transfer/test-config.yaml 5 files changed, 234 insertions(+), 216 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/19/19/3 -- To view, visit https://gerrit.asterisk.org/19 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.
Mark Michelson has uploaded a new patch set (#3). Change subject: Rewrite sip_attended_transfer test to stop failing. .. Rewrite sip_attended_transfer test to stop failing. The sip_attended_transfer test has been bouncing for a while. There are two major fixes introduced here to prevent the bouncing. First, by converting to using the testsuite's PJSUA module, we no longer are using the native Python threading library. Instead, we are using a method that works better with the Twisted framework. Second, the test is more strict about when the transfer may be performed. The previous test would attempt the transfer when Asterisk reported that the call was bridged. The problem is that Asterisk may report the call as bridged before PJSUA has properly processed the 200 OK that Asterisk has sent to it. By waiting, we can be sure that all parties are prepared when the transfer is attempted. The test has also been rewritten to only work with Asterisk 12+. A new separate test will be written to work on Asterisk 11. This helps the code to be a little less cluttered. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Address review feedback from Ashley * Two CallCallback classes have been combined into one * Bridge state has been factored into a minimal class * Unnecessary checks of test state have been removed. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e --- A tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/configs/ast1/extensions.conf M tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf D tests/channels/SIP/sip_attended_transfer/run-test M tests/channels/SIP/sip_attended_transfer/test-config.yaml 5 files changed, 234 insertions(+), 216 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/19/19/3 -- To view, visit https://gerrit.asterisk.org/19 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 3 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: rest_api/channels/snoop_spy: Stop test on bridge destruction
Mark Michelson has posted comments on this change. Change subject: rest_api/channels/snoop_spy: Stop test on bridge destruction .. Patch Set 1: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/21 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I373981f81ca455743bbf2371f07b380d013cd12b Gerrit-PatchSet: 1 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Matt Jordan mjor...@digium.com Gerrit-Reviewer: Joshua Colp jc...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
[asterisk-dev] Change in testsuite[master]: stasis: set a channel variable on websocket disconnect error
Mark Michelson has posted comments on this change.
Change subject: stasis: set a channel variable on websocket disconnect error
..
Patch Set 1:
(10 comments)
https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/ari_client.py
File tests/rest_api/applications/stasisstatus/ari_client.py:
Line 148: def on_channelcreated(self, message):
Unless I'm missing something, this, plus some of the other on_* functions
below, don't appear to be called during these tests. Can they be removed?
https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/observable_object.py
File tests/rest_api/applications/stasisstatus/observable_object.py:
Line 84: del self.__registrar[event][:]
I'm not 100% sure on this, but since you are using [:], that will return a copy
of the list at self.__registrar[event] rather than a reference to the actual
list, meaning that you aren't actually deleting the list at
self.__registrar[event]
I think that
del self.__registrar[event]
is actually what's wanted here.
Line 108: error += 'for event [{0}]; [Observers] is
None.'.format(event)
Nitpick: Start this string with a space, otherwise your error message will have
Could not register observersfor event...
Line 122: if self.__registrar[event] is None:
: msg += 'Instantiating the observers for event
{0}.'.format(event)
: LOGGER.debug(msg)
: self.__registrar[event] = list()
I may be misinterpreting your intent here, but I don't think this is going to
work how you expect it to. I've interpreted this block of code to mean that if
self.__registrar does not currently have the event key within it, then this
corrects the problem by setting self.__registrars[event] to be an empty list.
The problem with this is that if the key does not exist in self.__registrars,
then the if statement will throw a KeyError, not return None. You would need to
go with:
if event not in self.__registrar:
instead. Now, if you actually are trying to see if the value stored at the
event key in self.__registrar is None, then feel free to ignore this finding
completely.
Line 132: if self.__suspended 0:
: self.__suspended = 0
I don't imagine you actually want self.__suspended going less than 0, so this
should probably be = instead of
Line 143: def __validate(self, **kwargs):
This method feels a bit over-defensive. It's only ever called from two places,
and it has a single string passed to it. The method could be reduced down to
def __validate(self, event):
return True if event in self.__registrar else False
https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/run-test
File tests/rest_api/applications/stasisstatus/run-test:
Line 20: from stasisstatus.test_scenario_factory import build_scenarios
Hm, I don't see test_scenario_factory in this diff. Is it missing from the diff?
https://gerrit.asterisk.org/#/c/18/1/tests/rest_api/applications/stasisstatus/test_case.py
File tests/rest_api/applications/stasisstatus/test_case.py:
Line 64: TestCase.ami_connect(self, ami)
Calling TestCase.ami_connect() isn't necessary since ami_connect is a virtual
method in the first place. It's designed for you to just do what you need to do
for your derived class and be on your way.
Line 126: TestCase.run(self)
:
: self.create_ami_factory()
I think that TestCase.run(self) will already create the AMI factory, so you
don't need to do it yourself here.
In fact, I don't think it's really necessary to override this method since It's
just doing the same thing as the parent (plus printing a debug message)
Line 142: TestCase.stop_reactor(self)
This notation is a bit odd. Why not just self.stop_reactor()?
--
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Gerrit-MessageType: comment
Gerrit-Change-Id: I0f7dadfd429bd30e9f07a531f47884d8c923fc13
Gerrit-PatchSet: 1
Gerrit-Project: testsuite
Gerrit-Branch: master
Gerrit-Owner: Ashley Sanders asand...@digium.com
Gerrit-Reviewer: Mark Michelson mmichel...@digium.com
Gerrit-HasComments: Yes
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[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.
Mark Michelson has posted comments on this change. Change subject: Rewrite sip_attended_transfer test to stop failing. .. Patch Set 2: (6 comments) Thanks for the review! https://gerrit.asterisk.org/#/c/19/2/tests/channels/SIP/sip_attended_transfer/attended_transfer.py File tests/channels/SIP/sip_attended_transfer/attended_transfer.py: Line 36: class BobCallCallback(pj.CallCallback): Just as something to mention, BobCallCallback and CarolCallCallback can be Sounds good to me. Line 86: self.bridge1 = None Another fyi - The bridge1 and bridge2 could be made into a class and you co I like it. Line 121: self.call_carol() I think that you are calling this twice, once in the BobCallCallback.on_sta Well spotted! But, this is actually done intentionally. The reason is that we cannot guarantee the order of events. Bob's state may change to CONFIRMED before there are two channels in the Asterisk bridge, or it may happen the other way around. With this setup, they both attempt to call into the call_carol() function, and the call_carol() function will simply return early if the state is not such that calling Carol makes sense. Line 132: self.transfer_call() Here, too, this seems to be called twice; once in the CarolCallCallback.on_ And here it's the same deal as with your previous observation. Line 154: if (self.state == BOB_CALLED and self.bridge1_bridged and I don't think you need the BOB_CALLED state; if bob's call is up, then you This is a safeguard to ensure that we don't attempt to call Carol in a later stage of the test, say, after we've already performed the transfer. Looking again, I bet I could remove the check of the state from this function; however, the state itself as a class member is still necessary. The flow for a transfer goes as follows: Alice calls Bob, and they enter bridge 1. Alice calls Carol, and they enter bridge 2. Alice performs the transfer. Alice leaves both bridge 1 and 2. Now, the transfer code may move Bob out of bridge 1 and into bridge 2, or it may move Carol out of bridge 2 and into bridge 1. In either case, we detect the bridged state the same way as the original bridges with Alice: a BridgeEnter event with 2 channels in it. By maintaining the state of the test, we can determine whether a BridgeEnter with 2 channels means to continue on to the next state, or whether it means to hang up the calls because the test is complete. We also can detect if we get unexpected events and fail the test, as well. Line 162: if (self.state == CAROL_CALLED and self.bridge2_bridged and I don't think you need the CAROL_CALLED state; if carol's call is up, then See my reply about the BOB_CALLED state. -- To view, visit https://gerrit.asterisk.org/19 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Ashley Sanders asand...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: Yes -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.
Mark Michelson has uploaded a new patch set (#2). Change subject: Rewrite sip_attended_transfer test to stop failing. .. Rewrite sip_attended_transfer test to stop failing. The sip_attended_transfer test has been bouncing for a while. There are two major fixes introduced here to prevent the bouncing. First, by converting to using the testsuite's PJSUA module, we no longer are using the native Python threading library. Instead, we are using a method that works better with the Twisted framework. Second, the test is more strict about when the transfer may be performed. The previous test would attempt the transfer when Asterisk reported that the call was bridged. The problem is that Asterisk may report the call as bridged before PJSUA has properly processed the 200 OK that Asterisk has sent to it. By waiting, we can be sure that all parties are prepared when the transfer is attempted. The test has also been rewritten to only work with Asterisk 12+. A new separate test will be written to work on Asterisk 11. This helps the code to be a little less cluttered. Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e --- A tests/channels/SIP/sip_attended_transfer/attended_transfer.py M tests/channels/SIP/sip_attended_transfer/configs/ast1/extensions.conf M tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf D tests/channels/SIP/sip_attended_transfer/run-test M tests/channels/SIP/sip_attended_transfer/test-config.yaml 5 files changed, 242 insertions(+), 216 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/19/19/2 -- To view, visit https://gerrit.asterisk.org/19 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: Rewrite sip_attended_transfer test to stop failing.
Mark Michelson has uploaded a new change for review.
https://gerrit.asterisk.org/19
Change subject: Rewrite sip_attended_transfer test to stop failing.
..
Rewrite sip_attended_transfer test to stop failing.
The sip_attended_transfer test has been bouncing for a while. There
are two major fixes introduced here to prevent the bouncing.
First, by converting to using the testsuite's PJSUA module, we no
longer are using the native Python threading library. Instead, we
are using a method that works better with the Twisted framework.
Second, the test is more strict about when the transfer may be
performed. The previous test would attempt the transfer when Asterisk
reported that the call was bridged. The problem is that Asterisk may
report the call as bridged before PJSUA has properly processed the
200 OK that Asterisk has sent to it. By waiting, we can be sure that
all parties are prepared when the transfer is attempted.
The test has also been rewritten to only work with Asterisk 12+. A
new separate test will be written to work on Asterisk 11. This helps
the code to be a little less cluttered.
Change-Id: I1676801d90bcafc28ba25e8b6889f40ab08cc90e
---
A tests/channels/SIP/sip_attended_transfer/attended_transfer.py
M tests/channels/SIP/sip_attended_transfer/configs/ast1/extensions.conf
M tests/channels/SIP/sip_attended_transfer/configs/ast1/sip.conf
D tests/channels/SIP/sip_attended_transfer/run-test
M tests/channels/SIP/sip_attended_transfer/test-config.yaml
5 files changed, 242 insertions(+), 216 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/19/19/1
diff --git a/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
b/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
new file mode 100644
index 000..b925730
--- /dev/null
+++ b/tests/channels/SIP/sip_attended_transfer/attended_transfer.py
@@ -0,0 +1,191 @@
+
+Copyright (C) 2015, Digium, Inc.
+Mark Michelson mmichel...@digium.com
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+
+
+import logging
+import pjsua as pj
+
+from twisted.internet import reactor
+
+LOGGER = logging.getLogger(__name__)
+
+INIT = 0
+BOB_CALLED = 1
+CAROL_CALLED = 2
+TRANSFERRED = 3
+
+
+class TransferAccountCallback(pj.AccountCallback):
+'''Generic Account callback for Bob and Carol.
+
+The sole purpose of this callback is to auto-answer
+incoming calls
+'''
+
+def __init__(self, account):
+pj.AccountCallback.__init__(self, account)
+
+def on_incoming_call(self, call):
+call.answer(200)
+
+
+class BobCallCallback(pj.CallCallback):
+'''Call Callback used for Alice's call to Bob
+
+When we get told the call state is CONFIRMED, we signal
+to the test that the call is answered
+'''
+
+def __init__(self, call, transfer_object):
+pj.CallCallback.__init__(self, call)
+self.transfer_object = transfer_object
+
+def on_state(self):
+if self.call.info().state == pj.CallState.CONFIRMED:
+reactor.callFromThread(self.transfer_object.bob_call_answered)
+
+
+class CarolCallCallback(pj.CallCallback):
+'''Call Callback for Alice's call to Carol
+
+When we get told the call state is CONFIRMED we signal to the test that the
+call has been answered. This is very important, because if we attempt to
+perform the transfer before PJSUA has set the call state to CONFIRMED, then
+the REFER request that PJSUA sends will have a blank to-tag in the Refer-To
+header. Waiting ensures that all information is present in the REFER
+request.
+'''
+
+def __init__(self, call, transfer_object):
+pj.CallCallback.__init__(self, call)
+self.transfer_object = transfer_object
+
+def on_state(self):
+if self.call.info().state == pj.CallState.CONFIRMED:
+reactor.callFromThread(self.transfer_object.carol_call_answered)
+
+
+class Transfer(object):
+'''Controller for attended transfer test
+
+This contains all the methods for advancing the test, such as placing calls
+and performing transfers. It also has several state variables that help to
+determine the proper timing for performing actions.
+'''
+
+def __init__(self, test_object, accounts):
+super(Transfer, self).__init__()
+self.ami = test_object.ami[0]
+self.ami.registerEvent('BridgeCreate', self.bridge_create)
+self.ami.registerEvent('BridgeEnter', self.bridge_enter)
+
+self.bridge1 = None
+self.bridge2 = None
+self.bridge1_bridged = False
+self.bridge2_bridged = False
+self.bob_call_up = False
+self.carol_call_up = False
+
+bob = accounts.get('bob').account
+bob.set_callback(TransferAccountCallback(bob))
+
+carol = accounts.get('carol').account
+
[asterisk-dev] Change in testsuite[master]: Add SIP attended transfer for Asterisk 11.
Mark Michelson has uploaded a new change for review.
https://gerrit.asterisk.org/20
Change subject: Add SIP attended transfer for Asterisk 11.
..
Add SIP attended transfer for Asterisk 11.
The structure of this is similar to the new Asterisk 12+ SIP attended
transfer test, except that the method of determining if channels are
bridged is different.
Like the 12+ version, the two major changes to the original test are
the use of the PJSUA testsuite module and waiting to move forward
with test actions until PJSUA has informed us of state changes.
The logic used for determining bridgedness is the same as was used
in the sip_attended_transfer test that is being replaced.
Change-Id: I48c7b6a9298552aa756d0c2f26afbd6a96d553b5
---
A tests/channels/SIP/sip_attended_transfer_11/attended_transfer.py
A tests/channels/SIP/sip_attended_transfer_11/configs/ast1/extensions.conf
A tests/channels/SIP/sip_attended_transfer_11/configs/ast1/sip.conf
A tests/channels/SIP/sip_attended_transfer_11/test-config.yaml
M tests/channels/SIP/tests.yaml
5 files changed, 273 insertions(+), 0 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/20/20/1
diff --git a/tests/channels/SIP/sip_attended_transfer_11/attended_transfer.py
b/tests/channels/SIP/sip_attended_transfer_11/attended_transfer.py
new file mode 100644
index 000..5f22af4
--- /dev/null
+++ b/tests/channels/SIP/sip_attended_transfer_11/attended_transfer.py
@@ -0,0 +1,187 @@
+
+Copyright (C) 2015, Digium, Inc.
+Mark Michelson mmichel...@digium.com
+
+This program is free software, distributed under the terms of
+the GNU General Public License Version 2.
+
+
+import logging
+import pjsua as pj
+
+from twisted.internet import reactor
+
+LOGGER = logging.getLogger(__name__)
+
+INIT = 0
+BOB_CALLED = 1
+CAROL_CALLED = 2
+TRANSFERRED = 3
+
+
+class TransferAccountCallback(pj.AccountCallback):
+'''Generic Account callback for Bob and Carol.
+
+The sole purpose of this callback is to auto-answer
+incoming calls
+'''
+
+def __init__(self, account):
+pj.AccountCallback.__init__(self, account)
+
+def on_incoming_call(self, call):
+call.answer(200)
+
+
+class BobCallCallback(pj.CallCallback):
+'''Call Callback used for Alice's call to Bob
+
+When we get told the call state is CONFIRMED, we signal
+to the test that the call is answered
+'''
+
+def __init__(self, call, transfer_object):
+pj.CallCallback.__init__(self, call)
+self.transfer_object = transfer_object
+
+def on_state(self):
+if self.call.info().state == pj.CallState.CONFIRMED:
+reactor.callFromThread(self.transfer_object.bob_call_answered)
+
+
+class CarolCallCallback(pj.CallCallback):
+'''Call Callback for Alice's call to Carol
+
+When we get told the call state is CONFIRMED we signal to the test that the
+call has been answered. This is very important, because if we attempt to
+perform the transfer before PJSUA has set the call state to CONFIRMED, then
+the REFER request that PJSUA sends will have a blank to-tag in the Refer-To
+header. Waiting ensures that all information is present in the REFER
+request.
+'''
+
+def __init__(self, call, transfer_object):
+pj.CallCallback.__init__(self, call)
+self.transfer_object = transfer_object
+
+def on_state(self):
+if self.call.info().state == pj.CallState.CONFIRMED:
+reactor.callFromThread(self.transfer_object.carol_call_answered)
+
+
+class Transfer(object):
+'''Controller for attended transfer test
+
+This contains all the methods for advancing the test, such as placing calls
+and performing transfers. It also has several state variables that help to
+determine the proper timing for performing actions.
+'''
+
+def __init__(self, test_object, accounts):
+super(Transfer, self).__init__()
+self.ami = test_object.ami[0]
+self.ami.registerEvent('Bridge', self.bridge)
+self.ami.registerEvent('VarSet', self.bridge_peer)
+
+self.chans = []
+self.final_bridge = 0
+self.bob_call_up = False
+self.carol_call_up = False
+self.bridge1_bridged = False
+self.bridge2_bridged = False
+
+bob = accounts.get('bob').account
+bob.set_callback(TransferAccountCallback(bob))
+
+carol = accounts.get('carol').account
+carol.set_callback(TransferAccountCallback(carol))
+
+self.alice = accounts.get('alice').account
+self.call_to_bob = None
+self.call_to_carol = None
+
+self.test_object = test_object
+self.state = INIT
+
+def bridge(self, ami, event):
+if event['channel2'] in self.chans:
+return
+
+self.chans.append(event['channel2'])
+numchans = len(self.chans)
+if numchans == 1:
+self.bridge1_bridged
[asterisk-dev] Change in testsuite[master]: PJSIP: Added test to ensure retransmissions are not handled.
Mark Michelson has uploaded a new patch set (#2). Change subject: PJSIP: Added test to ensure retransmissions are not handled. .. PJSIP: Added test to ensure retransmissions are not handled. In this test, a SIPp scenario sends the exact same MESSAGE request to Asterisk twice. The test ensures that the dialplan is only called into a single time. Without the patch from https://reviewboard.asterisk.org/r/4532/ , this test fails because the UserEvent in the dialplan is sent twice. With that patch, this test succeeds. ASTERSIK-24920 Reported by Mark Michelson Change-Id: I524a3eb1cde4489d0ff9866913ae1be318c72115 --- A tests/channels/pjsip/message/message_retrans/configs/ast1/extensions.conf A tests/channels/pjsip/message/message_retrans/configs/ast1/pjsip.conf A tests/channels/pjsip/message/message_retrans/sipp/message_retrans.xml A tests/channels/pjsip/message/message_retrans/test-config.yaml M tests/channels/pjsip/message/tests.yaml 5 files changed, 94 insertions(+), 0 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/14/14/2 -- To view, visit https://gerrit.asterisk.org/14 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I524a3eb1cde4489d0ff9866913ae1be318c72115 Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in testsuite[master]: PJSIP: Added test to ensure retransmissions are not handled.
Mark Michelson has uploaded a new patch set (#2). Change subject: PJSIP: Added test to ensure retransmissions are not handled. .. PJSIP: Added test to ensure retransmissions are not handled. In this test, a SIPp scenario sends the exact same MESSAGE request to Asterisk twice. The test ensures that the dialplan is only called into a single time. Without the patch from https://reviewboard.asterisk.org/r/4532/ , this test fails because the UserEvent in the dialplan is sent twice. With that patch, this test succeeds. ASTERSIK-24920 Reported by Mark Michelson Change-Id: I524a3eb1cde4489d0ff9866913ae1be318c72115 --- A tests/channels/pjsip/message/message_retrans/configs/ast1/extensions.conf A tests/channels/pjsip/message/message_retrans/configs/ast1/pjsip.conf A tests/channels/pjsip/message/message_retrans/sipp/message_retrans.xml A tests/channels/pjsip/message/message_retrans/test-config.yaml M tests/channels/pjsip/message/tests.yaml 5 files changed, 94 insertions(+), 0 deletions(-) git pull ssh://gerrit.asterisk.org:29418/testsuite refs/changes/14/14/2 -- To view, visit https://gerrit.asterisk.org/14 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: newpatchset Gerrit-Change-Id: I524a3eb1cde4489d0ff9866913ae1be318c72115 Gerrit-PatchSet: 2 Gerrit-Project: testsuite Gerrit-Branch: master Gerrit-Owner: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-Reviewer: Matt Jordan mjor...@digium.com -- _ -- 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
[asterisk-dev] Change in repotools[master]: Add web proxy support to commit_msg.py
Mark Michelson has posted comments on this change. Change subject: Add web proxy support to commit_msg.py .. Patch Set 3: Code-Review+1 -- To view, visit https://gerrit.asterisk.org/13 To unsubscribe, visit https://gerrit.asterisk.org/settings Gerrit-MessageType: comment Gerrit-Change-Id: Ie71ccb85e09cce005847ce9bf70c603fbee3d58a Gerrit-PatchSet: 3 Gerrit-Project: repotools Gerrit-Branch: master Gerrit-Owner: Michael L. Young elgueromexic...@gmail.com Gerrit-Reviewer: Mark Michelson mmichel...@digium.com Gerrit-HasComments: No -- _ -- 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
