://reviewboard.asterisk.org/r/2964/diff/
Testing
---
Registered to an ITSP, placed an inbound call from them, confirmed matched
using line parameter.
Registered to a chan_sip instance, placed an inbound call from it, confirmed
matched using line parameter.
Thanks,
Joshua Colp
. Without patch that fixes issue it failed, with patch it passes.
Thanks,
Joshua Colp
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Diffs
-
/branches/12/res/res_pjsip_session.c 405776
Diff: https://reviewboard.asterisk.org/r/3140/diff/
Testing
---
Placed calls, confirmed that configured codecs are being sent now. Ran test and
confirmed it passes with change.
Thanks,
Joshua Colp
/attended_transfer/attended_transfer.py
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3145/diff/
Testing
---
Ran test, fails using branches. Applied patch for events and it passes!
Thanks,
Joshua Colp
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to respond to this email or join us on the #asterisk-dev IRC
channel. Have a spectacular day!
[1] https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite
[2] http://svn.asterisk.org/svn/asterisk/team/group/media_formats/
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445 Jan
://reviewboard.asterisk.org/r/3102 with a comment that it breaks NAT, as
it should then check against localnets to see if it should or shouldn't mangle
the address, that would be cool.
- Joshua Colp
On Jan. 6, 2014, 8:06 p.m., Joshua Colp wrote
that the correct IP address for each interface went into the message.
Some additional testing by others would be nice.
Thanks,
Joshua Colp
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On Jan. 22, 2014, 7:04 p.m
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- Joshua Colp
On Jan. 20, 2014, 10:03
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3139/diff/
Testing
---
Ran test. Without patch that fixes issue it failed, with patch it passes.
Thanks,
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now. Ran test and
confirmed it passes with change.
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5939 (SDP capability negotiation) which
changes things. Asterisk doesn't support this.
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...
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On Jan. 21, 2014, 12:48 p.m., Joshua Colp wrote:
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/asterisk/trunk/tests/rest_api/bridges/attended_transfer/attended_transfer.py
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3145/diff/
Testing
---
Ran test, fails using branches. Applied patch for events and it passes!
Thanks,
Joshua Colp
taking things into account
correctly.
Personally I am against optional encryption. Best effort encryption just
does not make sense to me.
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- Joshua Colp
On Jan. 28, 2014, 6:34 p.m
://reviewboard.asterisk.org/r/3168/#comment20216
You can't do this. Well, I mean, you can, it just won't persist. The
contact here is from the rdata pool and will end up getting reset later or just
get deallocated. You need to clone it using pjsip_hdr_clone from the dialog
pool.
- Joshua Colp
On Jan. 30, 2014
://reviewboard.asterisk.org/r/3168/#comment20217
You can also use pjsip_rdata_get_dlg here to retrieve a dialog, if one is
present, and have this work for non-INVITE stuff as well (just in case).
- Joshua Colp
On Jan. 30, 2014, 7:59 p.m., opticron wrote
://reviewboard.asterisk.org/r/3168/#comment20219
I would actually go so far as to do a URI comparison here... cause if this
happens multiple times you are going to keep growing the dialog pool usage.
- Joshua Colp
On Jan. 30, 2014, 8:27 p.m., opticron wrote
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- Joshua Colp
On Jan. 30, 2014, 9:38 p.m
at messages, profited.
Thanks,
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/configs/ast1/extensions.conf
PRE-CREATION
/asterisk/trunk/tests/rest_api/bridges/attended_transfer/attended_transfer.py
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3145/diff/
Testing
---
Ran test, fails using branches. Applied patch for events and it passes!
Thanks,
Joshua
/
Testing
---
Subscribed to an extension from a device that uses dialog-info+xml. Confirmed
NOTIFY messages going out contain the expected content.
Thanks,
Joshua Colp
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Testing
---
Ran the timing test using the various timing implementations on Linux and
confirmed they still work. I also placed some calls to a Playback and confirmed
it works.
Thanks,
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unit tests, all passed.
Note: I know AO2 throws a fit and it's because a container isn't getting
initialized. Getting said container initialized requires beginning the hacking
apart process.
Thanks,
Joshua Colp
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On Feb. 4, 2014, 3:16 p.m., Joshua Colp wrote
What happens if no name is specified? Should it return NULL or maybe place
a uuid in there instead?
- Joshua Colp
On Feb. 5, 2014, 5:08 p.m., George Joseph wrote:
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Testing
---
Ran the timing test using the various timing implementations on Linux and
confirmed they still work. I also placed some calls to a Playback and confirmed
it works.
Thanks,
Joshua Colp
://reviewboard.asterisk.org/r/3183/#comment20313
Should this actually just completely fail this? Otherwise you get a list
which doesn't contain all of your JSON variables. Which, if you experienced it
would cause you to scratch your head and spend time wondering what is going on.
- Joshua Colp
On Feb. 5
called now. The ao2 container has a reference
to the sorcery instance itself. You could do something where in
ast_sorcery_unref if it is the last reference it is unlinked from the instances
container.
- Joshua Colp
On Feb. 5, 2014, 8:48 p.m., George Joseph wrote
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- Joshua Colp
On Feb. 5, 2014, 9:10 p.m
.
/branches/12/res/ari/resource_channels.c
https://reviewboard.asterisk.org/r/3183/#comment20321
I'd throw out an error here if we got no variables.
- Joshua Colp
On Feb. 6, 2014, 2:51 p.m., Matt Jordan wrote:
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- Joshua Colp
On Feb. 5, 2014, 10:51 p.m
the
information to still exist for cases where you do want to get it.
- Joshua Colp
On Feb. 3, 2014, 8:18 p.m., Scott Griepentrog wrote:
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Mark this as being safe so if someone looks at it they don't go say what
- Joshua Colp
On Feb. 6, 2014, 5 p.m., George Joseph wrote:
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On Feb. 6, 2014, 3:52 p.m., Joshua Colp wrote:
While this works I'm not happy with pushing this down to such a low level.
What if in the future I want to filter something else?
What I'd really like to see is something on top which allows you to
arbitrarily filter anything
On Feb. 6, 2014, 3:52 p.m., Joshua Colp wrote:
While this works I'm not happy with pushing this down to such a low level.
What if in the future I want to filter something else?
What I'd really like to see is something on top which allows you to
arbitrarily filter anything
and get a warning,
but I still dislike the idea of hidden fields. ^_^
- Joshua Colp
On Feb. 6, 2014, 6:50 p.m., Scott Griepentrog wrote:
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://reviewboard.asterisk.org/r/3184/#comment20352
Er, do you mean you are unreffing an object that is already destroyed? If
so setting the variables to NULL will prevent that. Once this clarification is
provided this'll be good!
- Joshua Colp
On Feb. 6, 2014, 7:55 p.m., George Joseph wrote
://reviewboard.asterisk.org/r/3178/#review10770
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apart process.
Thanks,
Joshua Colp
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, 2014, 2:59 p.m., Joshua Colp wrote:
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(Updated Feb. 7, 2014, 2:59 p.m
/trunk/include/asterisk/timing.h 407425
Diff: https://reviewboard.asterisk.org/r/3175/diff/
Testing
---
Ran the timing test using the various timing implementations on Linux and
confirmed they still work. I also placed some calls to a Playback and confirmed
it works.
Thanks,
Joshua Colp
the hashtab usage and uses the heap for finding things.
Diffs
-
/trunk/main/sched.c 405157
Diff: https://reviewboard.asterisk.org/r/3199/diff/
Testing
---
Ran scheduler unit test and also forced the scheduler to get used for Playback.
Thanks,
Joshua Colp
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On Feb. 7, 2014, 2:59 p.m., Joshua Colp wrote:
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(Updated Feb. 7
and also forced the scheduler to get used for Playback.
Thanks,
Joshua Colp
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forced the scheduler to get used for Playback.
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a const value.
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string where contacts and matches are specified on
multiple lines?
Hilarity, that's why. Okay, not really. No immediate answer springs to mind.
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- Joshua Colp
On Feb. 12, 2014, 8:34 a.m
/app_alarmreceiver.c 408257
/team/file/media_formats-impl/apps/app_agent_pool.c 408257
Diff: https://reviewboard.asterisk.org/r/3234/diff/
Testing
---
Thanks,
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- Joshua Colp
On Feb. 20, 2014, 10:29
of metadata.
Diffs
-
/trunk/res/res_bucket_sounds.c PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3251/diff/
Testing
---
Executed CLI commands, output is as expected.
Thanks,
Joshua Colp
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- Joshua Colp
On Feb. 23, 2014, 5:48 a.m
address for each interface went into the message.
Some additional testing by others would be nice.
Thanks,
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negotiation?
- Joshua Colp
On Feb. 24, 2014, 6:36 p.m., Jonathan Rose wrote:
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getting
initialized. Getting said container initialized requires beginning the hacking
apart process.
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/app_amd.c 407953
/team/file/media_formats-impl/apps/app_alarmreceiver.c 407953
/team/file/media_formats-impl/apps/app_agent_pool.c 407953
Diff: https://reviewboard.asterisk.org/r/3234/diff/
Testing
---
Thanks,
Joshua Colp
Description
---
This change moves app_fax over to the new media formats API.
Diffs
-
/team/file/media_formats-impl/apps/app_fax.c 408263
Diff: https://reviewboard.asterisk.org/r/3265/diff/
Testing
---
It compiles.
Thanks,
Joshua Colp
-impl/codecs/codec_adpcm.c 407954
/team/file/media_formats-impl/codecs/codec_a_mu.c 407953
Diff: https://reviewboard.asterisk.org/r/3266/diff/
Testing
---
They compile, yay!
Thanks,
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On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote:
/branches/11/res/res_rtp_asterisk.c, lines 554-560
https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554
Agreed, and pjnath does not provide a mechanism to do just that without
destroying/re-creating as Matt
/res_pjsip_send_to_voicemail.c
https://reviewboard.asterisk.org/r/3245/#comment20572
Pfft, using RAII_VAR here is silly.
- Joshua Colp
On Feb. 20, 2014, 11:01 p.m., Kevin Harwell wrote:
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On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote:
/branches/11/res/res_rtp_asterisk.c, lines 554-560
https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554
Agreed, and pjnath does not provide a mechanism to do just that without
destroying/re-creating as Matt
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- Joshua Colp
On Feb. 25, 2014, 5:20 p.m
is removed from the INVITE session. Any attempts
to answer the INVITE session after this results in a crash as it requires
the INVITE transaction to exist. This change explicitly locks the dialog
and checks to ensure that the INVITE transaction exists before answering.
- Joshua
On Feb. 24, 2014, 10:51 p.m., Joshua Colp wrote:
/branches/11/res/res_rtp_asterisk.c, lines 554-560
https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554
Agreed, and pjnath does not provide a mechanism to do just that without
destroying/re-creating as Matt
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- Joshua Colp
On Feb. 21, 2014, 5 p.m
://reviewboard.asterisk.org/r/3267/#comment20588
Throw the channel name and endpoint in here for easier debugging further
down the road.
- Joshua Colp
On Feb. 26, 2014, 6:41 p.m., Scott Griepentrog wrote:
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- Joshua Colp
On Feb. 26, 2014, 6:52 p.m
and try to find other cases.
- Joshua Colp
On Feb. 27, 2014, 3:01 p.m., Michael Young wrote:
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://reviewboard.asterisk.org/r/3256/#comment20611
There's actually a bug here you didn't cause, but can fix!
If the component is RTP then rtp-s should be used, if the component is
RTCP then rtp-rtcp-s should be used.
Otherwise looks good!
- Joshua Colp
On Feb. 27, 2014, 4:38 p.m., Jonathan
On Feb. 27, 2014, 6:14 p.m., Joshua Colp wrote:
/branches/11/res/res_rtp_asterisk.c, line 1689
https://reviewboard.asterisk.org/r/3256/diff/2/?file=54884#file54884line1689
There's actually a bug here you didn't cause, but can fix!
If the component is RTP then rtp-s
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- Joshua Colp
On Feb. 27, 2014, 6:41 p.m
remote candidates into proposed and then moving them to active?
- Joshua Colp
On Feb. 27, 2014, 8:02 p.m., Jonathan Rose wrote:
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On Feb. 27, 2014, 4:26 p.m., Joshua Colp wrote:
This is fine but we should probably do an audit and try to find other cases.
Corey Farrell wrote:
If nobody is working on the audit of channel usage in dialplan functions
I can. Should I start a new JIRA ticket?
That'd be nifty
of Apache on port 80.
Instead of binding both to 0.0.0.0 you can bind them to two different
interface IP addresses if you want.
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commands:
pjsip set logger on
pjsip set logger on host/IP address
pjsip set logger off
You can also enable it at startup by using the debug option in the
global section of res_pjsip.conf
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Check
is behind NAT. They specify what
is local to it so SIP messages don't get rewritten, or if going outside
of that - they do get rewritten to the external address.
The options used when a remote endpoint is behind NAT are:
force_rport=yes
rtp_symmetric=yes
rewrite_contact=yes
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Digium
)? It's kind of a bummer to have phones
that think they are still registered, but you can't reach them.
Looked thru the literature, haven't seen anything juicy about
this.
OPTIONS functionality is controlled by specifying qualify_frequency on
the AOR.
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Digium, Inc. | Senior Software
://reviewboard.asterisk.org/r/3275/#comment20658
This will crash and left_candidate will never be non-NULL.
Take out that ao2_ref and this is good to go.
- Joshua Colp
On March 3, 2014, 5:01 p.m., Jonathan Rose wrote
, which of course fails.
Any ideas?
What's the exact configuration in use? Do you have a transport
explicitly specified for the endpoint? Doing so will currently cause it
to try to create a new connection [1].
[1] https://issues.asterisk.org/jira/browse/ASTERISK-22658
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- Joshua Colp
On March 3, 2014, 6:02 p.m
connection. What happens if you use rewrite_contact=yes on the endpoint?
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://reviewboard.asterisk.org/r/3102/#review11060
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On Feb. 23, 2014, 5:57 p.m., Joshua Colp wrote:
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Testing
---
Started recordings and confirmed target_uri is as expected.
Thanks,
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/recording/nominal/configs/ast1/extensions.conf
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3303/diff/
Testing
---
Ran test, it passes.
Thanks,
Joshua Colp
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On March 6, 2014, 1:38 a.m., Matt Jordan wrote:
branches/12/res/res_pjsip_endpoint_identifier_ip.c, lines 364-371
https://reviewboard.asterisk.org/r/3283/diff/1/?file=55030#file55030line364
Any reason you decided to move initialization of the members of the
struct to load_module?
with other stuff, it may not be. Can't know. The
set_extended usage has to go.
Create a structure to contain additional information, put stuff in that,
and pass it in.
- Joshua Colp
On March 3, 2014, 12:26 a.m., George Joseph wrote
/r/3305/#comment20740
Less RAII_VAR please. :P
- Joshua Colp
On March 5, 2014, 11:40 p.m., Mark Michelson wrote:
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://reviewboard.asterisk.org/r/3304/#comment20741
Remove.
- Joshua Colp
On March 5, 2014, 8:19 p.m., Jonathan Rose wrote:
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while here.
- Joshua Colp
On March 4, 2014, 10:56 p.m., Scott Griepentrog wrote:
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Ship it!
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- Joshua Colp
On March 6, 2014, 3:44 p.m
409828
/branches/12/include/asterisk/stasis_app_recording.h 409828
Diff: https://reviewboard.asterisk.org/r/3299/diff/
Testing
---
Started recordings and confirmed target_uri is as expected.
Thanks,
Joshua Colp
/tests/rest_api/recording/nominal/recording.py PRE-CREATION
/asterisk/trunk/tests/rest_api/recording/nominal/configs/ast1/extensions.conf
PRE-CREATION
Diff: https://reviewboard.asterisk.org/r/3303/diff/
Testing
---
Ran test, it passes.
Thanks,
Joshua Colp
throws a fit and it's because a container isn't getting
initialized. Getting said container initialized requires beginning the hacking
apart process.
Thanks,
Joshua Colp
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- Joshua Colp
On March 6, 2014, 9:51 p.m
407953
/team/file/media_formats-impl/apps/app_amd.c 407953
/team/file/media_formats-impl/apps/app_alarmreceiver.c 407953
/team/file/media_formats-impl/apps/app_agent_pool.c 407953
Diff: https://reviewboard.asterisk.org/r/3234/diff/
Testing
---
Thanks,
Joshua Colp
/codec_dahdi.c 407953
/team/file/media_formats-impl/codecs/codec_alaw.c 407953
/team/file/media_formats-impl/codecs/codec_adpcm.c 407954
/team/file/media_formats-impl/codecs/codec_a_mu.c 407953
Diff: https://reviewboard.asterisk.org/r/3266/diff/
Testing
---
They compile, yay!
Thanks,
Joshua
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It compiles.
Thanks,
Joshua Colp
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