/3550/diff/
Testing
---
Tested with cross compile for armv7hl platform (package paths specified) and
with native x86_64 compile (no package paths specified).
Thanks,
George Joseph
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Tests: 88 Passed: 87 Failed: 1
FAILED: tests/channels/pjsip/dialplan_functions/pjsip_endpoint
Not sure why the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
, visit:
https://reviewboard.asterisk.org/r/3576/#review12014
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On May 29, 2014, 6:53 p.m., George Joseph wrote:
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Thanks,
George Joseph
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tests/channels/pjsip/
Tests: 88 Passed: 87 Failed: 1
FAILED: tests/channels/pjsip/dialplan_functions/pjsip_endpoint
Not sure why the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
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On June 4, 2014, 7:33 p.m., George Joseph wrote:
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(Updated June 4, 2014, 7
the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
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On June 5, 2014, 4:38 p.m., George Joseph wrote:
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Not sure why the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
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/dialplan_functions/pjsip_endpoint
Not sure why the pjsip_endpoint function is failing but it's not this patch's
fault.
Thanks,
George Joseph
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asterisk
branches/12/main/astobj2_container.c 415336
Diff: https://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
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Right now, the non-ref debugging code in astobj2 is triggered by a mix of
AST_DEVMODE and AO2_DEBUG and both get set if you want to run the test
framework. I've noticed though that the inclusion of the debugging code
can actually hide problems as well as highlight them, especially related to
purposes.
On Mon, Jun 9, 2014 at 12:46 PM, Matthew Jordan mjor...@digium.com
wrote:
On Sun, Jun 8, 2014 at 10:03 AM, George Joseph
george.jos...@fairview5.com wrote:
Right now, the non-ref debugging code in astobj2 is triggered by a mix
of
AST_DEVMODE and AO2_DEBUG and both get set if you want
/build_tools/cflags.xml 415657
Diff: https://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
On Wed, Jun 11, 2014 at 12:24 PM, Mark Michelson mmichel...@digium.com
wrote:
Hi!
It has been brought to my attention that chan_pjsip does not have an
equivalent to chan_sip's usereqphone option. However, nobody here seems to
know how useful such an option actually would be. The result is,
.
Thanks,
George Joseph
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/extensions.lua.sample 416556
Diff: https://reviewboard.asterisk.org/r/3627/diff/
Testing
---
Made sure extensions.lua now loads correctly.
Thanks,
George Joseph
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://reviewboard.asterisk.org/r/3629/diff/
Testing
---
Checked that contexts created by both pbx_config and pbx_lua were properly
merged instead of pbx_config overwriting them.
Thanks,
George Joseph
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of pbx_config overwriting them.
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/acl.h 416729
Diff: https://reviewboard.asterisk.org/r/3652/diff/
Testing
---
Made sure both ipv4 and ipv6 addresses were formatted correctly.
Thanks,
George Joseph
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416729
branches/12/include/asterisk/netsock2.h 416729
branches/12/include/asterisk/acl.h 416729
Diff: https://reviewboard.asterisk.org/r/3652/diff/
Testing
---
Made sure both ipv4 and ipv6 addresses were formatted correctly.
Thanks,
George Joseph
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On June 19, 2014, 11:47 a.m., George Joseph wrote
,
George Joseph
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On June 10, 2014, 8:42 a.m., George Joseph wrote:
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https://reviewboard.asterisk.org/r/3593
://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
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://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
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://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
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/build_tools/cflags.xml 416802
Diff: https://reviewboard.asterisk.org/r/3593/diff/
Testing
---
Ran all the test framework tests plus the astobj2 test test.
Ran testsuite tests/channels/pjsip.
Thanks,
George Joseph
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On May 20, 2014, 3 p.m., George Joseph wrote
were added to the existing astobj2 test framework. All tests pass.
The testsuite is a little more challenging but I at least made sure that all
the test that passed before the change still passed after the change.
Thanks,
George Joseph
/3550/diff/
Testing
---
Tested with cross compile for armv7hl platform (package paths specified) and
with native x86_64 compile (no package paths specified).
Thanks,
George Joseph
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of _FORTIFY_SOURCE and -O were set.
Thanks,
George Joseph
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Testing
---
Tested compile both with and without the optimize flag set and insured that the
proper combination of _FORTIFY_SOURCE and -O were set.
Thanks,
George Joseph
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Ship it!
Ship It!
- George Joseph
On June 24, 2014, 12:28
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Ship it!
Tested on my Grandstreams. Ship It!
- George
.
The full Testsuite was run with the same number of tests passing that passed
before the patch.
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George Joseph
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On July 7, 2014, 10:43 p.m., George Joseph wrote
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George Joseph wrote:
An object can't use a list of containers as this would mean the container
could only be in one object's list at a time. I'd have to use a container of
containers which would be significant overhead. That's why I went with node
since a node can only be associated with one
.
George Joseph wrote:
An object can't use a list of containers as this would mean the container
could only be in one object's list at a time. I'd have to use a container of
containers which would be significant overhead. That's why I went with node
since a node can only be associated with one
.
George Joseph wrote:
An object can't use a list of containers as this would mean the container
could only be in one object's list at a time. I'd have to use a container of
containers which would be significant overhead. That's why I went with node
since a node can only be associated with one
if it happens to define the same context/exten as
pbx_config or pbx_ael it'll never be called because pbx_find_extension will
never search pbx_lua unless the global table is exhausted.
Thanks,
George Joseph
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the global table is exhausted.
Thanks,
George Joseph
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Diff: https://reviewboard.asterisk.org/r/3919/diff/
Testing
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Thanks,
George Joseph
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. If a dialplan developer
wants to debug their dialplan they still can.
Diffs (updated)
-
branches/12/funcs/func_config.c 421326
Diff: https://reviewboard.asterisk.org/r/3919/diff/
Testing
---
Thanks,
George Joseph
ERROR to DEBUG. If a dialplan developer
wants to debug their dialplan they still can.
Diffs
-
branches/12/funcs/func_config.c 421326
Diff: https://reviewboard.asterisk.org/r/3919/diff/
Testing
---
Thanks,
George Joseph
-insensitive since kick was already case-insensitive.
Diffs
-
branches/12/apps/app_confbridge.c 422052
Diff: https://reviewboard.asterisk.org/r/3944/diff/
Testing
---
Tested that both CLI and AMI handle 'all' as a channel target for mute and
unmute correctly.
Thanks,
George Joseph
On Aug. 25, 2014, 11:53 a.m., Matt Jordan wrote:
How is this different than the -l option?
Just to chip in... The -l option lists everything even though the test might
not actually get run because of some constraint. I use something like the
proposed dry-run so I can know how many tests
://reviewboard.asterisk.org/r/3944/diff/
Testing (updated)
---
Tested that both CLI and AMI handle 'all' and 'participants' as a channel
target for mute, unmute and kick correctly.
Thanks,
George Joseph
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://reviewboard.asterisk.org/r/3944/diff/
Testing
---
Tested that both CLI and AMI handle 'all' and 'participants' as a channel
target for mute, unmute and kick correctly.
Thanks,
George Joseph
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'all' and 'participants' as a channel
target for mute, unmute and kick correctly.
Thanks,
George Joseph
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' and 'participants' as a channel
target for mute, unmute and kick correctly.
Thanks,
George Joseph
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Testing
---
Tested that both CLI and AMI handle 'all' and 'participants' as a channel
target for mute, unmute and kick correctly.
Thanks,
George Joseph
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422108
Diff: https://reviewboard.asterisk.org/r/3950/diff/
Testing
---
Checked that the AMI events had properly set Admin flags.
All of the 11 eligible TestSuite confbridge tests passed although they don't
test for the existence of the new flag yet. Nothing broke at least.
Thanks,
George
they don't
test for the existence of the new flag yet. Nothing broke at least.
Thanks,
George Joseph
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On Aug. 27, 2014, 10:56 a.m., George Joseph wrote
eligible TestSuite confbridge tests passed although they don't
test for the existence of the new flag yet. Nothing broke at least.
Thanks,
George Joseph
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/3955/diff/
Testing
---
Looked at the output of a ConfbridgeList event to make sure Duration was being
set correctly and no other fields were affected.
Thanks,
George Joseph
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---
Verified that events to be filtered are indeed filtered and that events not
supposed to be filtered aren't.
I ran the manager TestSuite but 4 of 9 tests failed BEFORE my change. The same
4 failed after my change.
Thanks,
George Joseph
The last hurdle for me to adopt pjsip has been what to do with phoneprov
and now that chan_sip has been moved to extended support, this has come to
the front burner. I've been thinking of ways to make the existing
res_phoneprov configurable to use either the existing users.conf/sip.conf
.
Diffs (updated)
-
branches/12/apps/app_confbridge.c 422257
Diff: https://reviewboard.asterisk.org/r/3955/diff/
Testing
---
Looked at the output of a ConfbridgeList event to make sure Duration was being
set correctly and no other fields were affected.
Thanks,
George Joseph
tests failed BEFORE my change. The same
4 failed after my change.
Thanks,
George Joseph
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affected.
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George Joseph
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actually use it with Grandstream phones and everything worked exactly as
expected.
Thanks,
George Joseph
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and
PP_EACH_EXTENSION to make sure that all existing functionality was preserved.
I actually use it with Grandstream phones and everything worked exactly as
expected.
Thanks,
George Joseph
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templates.
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George Joseph
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/3970/diff/
Testing
---
I ran through several scenarios including the use of PP_EACH_USER and
PP_EACH_EXTENSION to make sure that all existing functionality was preserved.
I actually use it with Grandstream phones and everything worked exactly as
expected.
Thanks,
George Joseph
/res_pjsip_phoneprov_provider.c PRE-CREATION
branches/12/configs/pjsip.conf.sample 422737
Diff: https://reviewboard.asterisk.org/r/3976/diff/
Testing
---
I'm already starting to convert sip peers to pjsip endpoints with no change to
my Grandstream templates.
Thanks,
George Joseph
that no files are truncated if any of the
can't be written to.
Thanks,
George Joseph
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://reviewboard.asterisk.org/r/3986/diff/
Testing
---
Tested normal access and verified that no files are truncated if any of the
can't be written to.
Thanks,
George Joseph
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-
branches/1.8/main/config.c 422882
Diff: https://reviewboard.asterisk.org/r/3986/diff/
Testing
---
Tested normal access and verified that no files are truncated if any of the
can't be written to.
Thanks,
George Joseph
,
George Joseph
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On Sept. 10, 2014, 5:45 p.m., George Joseph wrote:
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---
(Updated
(updated)
-
branches/12/tests/test_strings.c 422963
branches/12/main/utils.c 422963
branches/12/include/asterisk/strings.h 422963
Diff: https://reviewboard.asterisk.org/r/3989/diff/
Testing
---
Thanks,
George Joseph
422963
branches/12/include/asterisk/strings.h 422963
Diff: https://reviewboard.asterisk.org/r/3989/diff/
Testing
---
Thanks,
George Joseph
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---
Tested with AMI UpdateConfig/newcat to make sure the proper value is returned
in both positive and negative cases.
Thanks,
George Joseph
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/include/asterisk/config.h 423127
Diff: https://reviewboard.asterisk.org/r/3993/diff/
Testing
---
Tested with AMI UpdateConfig/newcat to make sure the proper value is returned
in both positive and negative cases.
Thanks,
George Joseph
);
Diffs (updated)
-
branches/12/tests/test_strings.c 422963
branches/12/main/utils.c 422963
branches/12/include/asterisk/strings.h 422963
Diff: https://reviewboard.asterisk.org/r/3989/diff/
Testing
---
Thanks,
George Joseph
On Tue, Sep 16, 2014 at 1:48 PM, Matthew Jordan mjor...@digium.com wrote:
And there was much rejoicing
To summarize:
* A comparison of management platforms has been done. Barring a giant
catastrophe or some insane limitation, we're going to go simple here and
stick with gitolite. Reasoning
to make sure the error appears (or
doesn't) correctly and that ListItems is set correctly.
Thanks,
George Joseph
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On Sept. 16, 2014, 12:16 p.m., Scott Griepentrog wrote:
/branches/12/res/res_pjsip_endpoint_identifier_ip.c, lines 162-166
https://reviewboard.asterisk.org/r/3995/diff/1/?file=67358#file67358line162
The implementation of ast_append_ha supports comma separated values,
but
/asterisk/res_pjsip.h 423274
Diff: https://reviewboard.asterisk.org/r/3998/diff/
Testing
---
Checked the output of PJSIPShowEndpoint to make sure the error appears (or
doesn't) correctly and that ListItems is set correctly.
Thanks,
George Joseph
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On Sept. 17, 2014, 5:43 p.m., George Joseph wrote
branches/1.8/main/config.c 423127
branches/1.8/include/asterisk/config.h 423127
Diff: https://reviewboard.asterisk.org/r/3993/diff/
Testing
---
Tested with AMI UpdateConfig/newcat to make sure the proper value is returned
in both positive and negative cases.
Thanks,
George Joseph
}
\endcode
*/
char *ast_strsep(char **s, const char sep, uint32_t flags);
Diffs
-
branches/12/tests/test_strings.c 422963
branches/12/main/utils.c 422963
branches/12/include/asterisk/strings.h 422963
Diff: https://reviewboard.asterisk.org/r/3989/diff/
Testing
---
Thanks,
George
actually use it with Grandstream phones and everything worked exactly as
expected.
Thanks,
George Joseph
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Testing
---
I'm already starting to convert sip peers to pjsip endpoints with no change to
my Grandstream templates.
Thanks,
George Joseph
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On Sept. 18, 2014, 6:01 p.m., George Joseph wrote:
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was preserved.
I actually use it with Grandstream phones and everything worked exactly as
expected.
Thanks,
George Joseph
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:21 p.m., George Joseph wrote:
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(Updated Sept. 18, 2014, 3:21 p.m
/r/3976/#review13359
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On Sept. 18, 2014, 3:21 p.m., George Joseph wrote:
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Testing
---
I'm already starting to convert sip peers to pjsip endpoints with no change to
my Grandstream templates.
Thanks,
George Joseph
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In the process of getting my GUI and real customers up on PJSIP I've come
across a few things that could work better for me. One of them is the
configuration process for ITSP trunks, specifically those that require
registration and have more than 1 server.
In a simple single-server scenario, a
On Sat, Sep 20, 2014 at 10:33 AM, Joshua Colp jc...@digium.com wrote:
George Joseph wrote:
snip
My proposal is to allow registration/server_uri to be specified as a
comma separated list or to be specified multiple times just like
aor/contact and identify/match. This would allow us
On Sat, Sep 20, 2014 at 11:21 AM, Joshua Colp jc...@digium.com wrote:
George Joseph wrote:
snip
5. The idea of higher level concept configuration has been thrown
around as something to make this easier. I personally think this
sort of thing belongs there. A type=trunk, itsp
On Sat, Sep 20, 2014 at 12:06 PM, George Joseph george.jos...@fairview5.com
wrote:
On Sat, Sep 20, 2014 at 11:21 AM, Joshua Colp jc...@digium.com wrote:
George Joseph wrote:
snip
5. The idea of higher level concept configuration has been thrown
around as something to make
On Sat, Sep 20, 2014 at 1:10 PM, Joshua Colp jc...@digium.com wrote:
George Joseph wrote:
snip
Or separate objects from a config file perspective but implemented in
pjsip_configuration with endpoint.
Completely separate. Mixing the two defeats the purpose of having a clear
boundary
On Sat, Sep 20, 2014 at 1:35 PM, Joshua Colp jc...@digium.com wrote:
George Joseph wrote:
On Sat, Sep 20, 2014 at 1:10 PM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
George Joseph wrote:
snip
Or separate objects from a config file perspective
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