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- Joshua Colp
On July 24, 2014, 3:58 p.m
modifications.
Thoughts?
- Joshua Colp
On July 25, 2014, 2:20 p.m., rnewton wrote:
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-CREATION
Diff: https://reviewboard.asterisk.org/r/3780/diff/
Testing
---
Set up two Asterisk instances, configured both sides to publish to eachother,
made calls and manipulated voicemail. Watched PUBLISH messages go between them
and state change.
Thanks,
Joshua Colp
sscanf this.
- Joshua Colp
On July 25, 2014, 2:47 p.m., Michael Young wrote:
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this review or
the commit you won't know it's possible.
- Joshua Colp
On July 21, 2014, 9:54 p.m., Jason Parker wrote:
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- Joshua Colp
On July 27, 2014, 3:39 p.m
,
Joshua Colp
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-CREATION
Diff: https://reviewboard.asterisk.org/r/3780/diff/
Testing
---
Set up two Asterisk instances, configured both sides to publish to eachother,
made calls and manipulated voicemail. Watched PUBLISH messages go between them
and state change.
Thanks,
Joshua Colp
, it passes. Sabotaged test, it fails.
Thanks,
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them
and state change.
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Joshua Colp
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Olle E. Johansson wrote:
On 31 Jul 2014, at 17:28, Joshua Colp reviewbo...@asterisk.org
mailto:reviewbo...@asterisk.org wrote:
This adds two PJSIP modules which add outbound PUBLISH support and an
'asterisk' event type.
I don't think it's a good idea to mix different events in one event
digits before execute ast_bridge_call.
I made something wrong but I don't know :-) thank to help me
Have you waited for frames on the channel using something like
ast_waitfor? If you try to read when nothing is available on the channel
you will see the message above.
Cheers,
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Digium
so we can get a
consensus on the exact behavior that should occur here. I realize that this
solves the problem for you but I want to make sure everyone is on the same page
and in agreement for the resulting behavior.
- Joshua Colp
On July 30, 2014, 11:19 p.m., Corey Farrell wrote
it will unbridge/return
when it receives DTMF on the respective channel you want.
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simultaneously, even with
locking internally. It won't work as you expect/want as only a single
thread can service a channel. You would essentially end up with two
threads fighting over the frames, with some going to one thread and some
going to the other.
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- Joshua Colp
On Aug. 4, 2014, 9:39 p.m
and 12-12
that makes it so we need to remove the newly added ones?
- Joshua Colp
On Aug. 4, 2014, 10:47 p.m., rmudgett wrote:
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On Aug. 5, 2014, 1:03 p.m., Joshua Colp wrote:
Can you elaborate on what actually happens here between 11-12 and 12-12
that makes it so we need to remove the newly added ones?
rmudgett wrote:
The format compatibility bits need to remain frozen to new codecs because
chan_iax2
/#review12996
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On Aug. 3, 2014, 10:58 p.m., Joshua Colp wrote:
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, it passes. Sabotaged test, it fails.
Thanks,
Joshua Colp
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-CREATION
Diff: https://reviewboard.asterisk.org/r/3780/diff/
Testing
---
Set up two Asterisk instances, configured both sides to publish to eachother,
made calls and manipulated voicemail. Watched PUBLISH messages go between them
and state change.
Thanks,
Joshua Colp
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On Aug. 5, 2014, 5:37 p.m., Joshua Colp wrote
-CREATION
Diff: https://reviewboard.asterisk.org/r/3780/diff/
Testing
---
Set up two Asterisk instances, configured both sides to publish to eachother,
made calls and manipulated voicemail. Watched PUBLISH messages go between them
and state change.
Thanks,
Joshua Colp
5316
Diff: https://reviewboard.asterisk.org/r/3869/diff/
Testing
---
Executed test, it passes. Sabotaged test, it fails.
Thanks,
Joshua Colp
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PUBLISH messages go between them
and state change.
Thanks,
Joshua Colp
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: https://reviewboard.asterisk.org/r/3914/diff/
Testing
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Thanks,
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. This
change ensures they are present in the resulting HTTP GET request.
Diffs
-
/branches/13/res/res_http_websocket.c 421123
Diff: https://reviewboard.asterisk.org/r/3914/diff/
Testing
---
Thanks,
Joshua Colp
://reviewboard.asterisk.org/r/3917/#comment23547
Add a comment explaining why this is being done.
- Joshua Colp
On Aug. 15, 2014, 3:53 p.m., Jonathan Rose wrote:
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strcpy with a comment that it's safe is fine here
- Joshua Colp
On Aug. 6, 2014, 6:38 p.m., Michael Young wrote:
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Ship it!
This applies to 1.8+
- Joshua Colp
On Aug. 7
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Ship it!
Ship It!
- Joshua Colp
On Aug. 13, 2014, 10:04
On Aug. 18, 2014, 2:24 p.m., Joshua Colp wrote:
/branches/12/res/res_stasis.c, lines 1330-1333
https://reviewboard.asterisk.org/r/3917/diff/1/?file=66549#file66549line1330
Add a comment explaining why this is being done.
Jonathan Rose wrote:
Honestly, I don't believe
://reviewboard.asterisk.org/r/3917/#comment23552
a new PBX
- Joshua Colp
On Aug. 18, 2014, 3:27 p.m., Jonathan Rose wrote:
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- Joshua Colp
On Aug. 15, 2014, 1:55 a.m
expected to be
invoked at #3 - do you think this is possible at all?
- Joshua Colp
On Aug. 24, 2014, 3:52 p.m., Mark Michelson wrote:
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On Aug. 24, 2014, 3:56 p.m., Joshua Colp wrote:
My only concern with this is that we somehow have code which expected to be
invoked at #3 - do you think this is possible at all?
It may also be useful to document your flow discovery above within the code
somewhere for future reference
On Aug. 24, 2014, 3:56 p.m., Joshua Colp wrote:
My only concern with this is that we somehow have code which expected to be
invoked at #3 - do you think this is possible at all?
Joshua Colp wrote:
It may also be useful to document your flow discovery above within the
code
, and also run
through all the PJSIP tests. Do that and this is good to go.
- Joshua Colp
On Aug. 24, 2014, 11:37 p.m., Mark Michelson wrote:
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On Aug. 25, 2014, 1:12 p.m., Joshua Colp wrote:
Review your doxygen because I think you are repeating yourself, and also
run through all the PJSIP tests. Do that and this is good to go.
Mark Michelson wrote:
I don't see the doxygen problem. One states that on transaction state
/res_pjsip_session.h
https://reviewboard.asterisk.org/r/3930/#comment23664
Document the default value for this.
- Joshua Colp
On Aug. 26, 2014, 10:44 p.m., Mark Michelson wrote:
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=identify section.
In the future for this type of help the asterisk-users mailing list
would be more appropriate.
[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
Cheers,
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this or something similar in asterisk-13.0.0-beta1.
This issue was reported here[1] and has been fixed in SVN. The fix will
be in the next releases.
Cheers,
[1] https://issues.asterisk.org/jira/browse/ASTERISK-24234
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and sent it to a Playback. Confirmed
that RTP was sent to the provided IP address/port with the given format.
Thanks,
Joshua Colp
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/asterisk/rtp_engine.h 422835
Diff: https://reviewboard.asterisk.org/r/3982/diff/
Testing
---
Sabotaged the code so the only candidates I sent were of the relay type.
Confirmed bidirectional media flow using the TURN relay.
Thanks,
Joshua Colp
Johann Steinwendtner wrote:
On 2014-09-07 17:07, Joshua Colp wrote:
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Testing
Originated a call to a UnicastRTP channel and sent it to a Playback.
Confirmed that RTP was sent to the provided
/res_rtp_asterisk.c 422898
/branches/13/include/asterisk/rtp_engine.h 422898
Diff: https://reviewboard.asterisk.org/r/3982/diff/
Testing
---
Sabotaged the code so the only candidates I sent were of the relay type.
Confirmed bidirectional media flow using the TURN relay.
Thanks,
Joshua Colp
/3985/#comment23754
It's still possible for this to occur under extreme circumstances (out of
memory).
- Joshua Colp
On Sept. 9, 2014, 5:04 p.m., Jonathan Rose wrote:
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- Joshua Colp
On Sept. 10, 2014, 1:34
Magic number. Define this somewhere.
/branches/13/res/res_pjsip/pjsip_options.c
https://reviewboard.asterisk.org/r/3954/#comment23758
Same may or may not be true thing applies from above.
- Joshua Colp
On Sept. 4, 2014, 11:41 p.m., rmudgett wrote
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Ship it!
Ship It!
- Joshua Colp
On Sept. 4, 2014, 8:11 p.m
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- Joshua Colp
On Sept. 9, 2014, 5:04 p.m., Jonathan Rose
On Sept. 10, 2014, 11:51 a.m., Joshua Colp wrote:
/branches/13/res/res_pjsip.c, lines 2478-2480
https://reviewboard.asterisk.org/r/3954/diff/3/?file=67221#file67221line2478
I'm hesitant to say that this will always be true. I fear that due to
the asynchronous nature of things
://reviewboard.asterisk.org/r/3988/#comment23764
Test this and make sure that the PJSIP digest authentication doesn't freak.
:P
- Joshua Colp
On Sept. 10, 2014, 5:53 p.m., Sean Bright wrote:
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Olivier wrote:
snip
Resource List Subscriptions ?
This is definitely something I've wanted to test but forgot to add it to
the list. Now added!
Thanks,
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Check us out
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Ship It!
- Joshua Colp
On Sept. 10, 2014, 9:31
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Ship it!
Ship It!
- Joshua Colp
On Sept. 10, 2014, 3:56
/r/3954/#comment23767
Looking at the pjproject code why wouldn't the following return values
cover these cases for when the callback is not called:
PJ_EINVAL
PJ_EINVALIDOP
PJ_ENOMEM
Handling those as special would reduce the memory leak possibility.
- Joshua Colp
/chan_multicast_rtp.c 422835
Diff: https://reviewboard.asterisk.org/r/3981/diff/
Testing
---
Originated a call to a UnicastRTP channel and sent it to a Playback. Confirmed
that RTP was sent to the provided IP address/port with the given format.
Thanks,
Joshua Colp
://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
Optimistic Off Success (Encrypted)
Mandatory Optimistic On Success (Encrypted)
Optional Optimistic Off Success (Encrypted)
Optional Optimistic On Success (Encrypted)
Thanks,
Joshua Colp
it. The UDPTL and RTP stacks don't share
sockets and the RTP side keeps the port open while UDPTL is going on so
even if the UDPTL side wanted to use the same port it couldn't bind to it.
Can you provide the trace of this with SDP and logging?
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Digium, Inc. | Senior Software Developer
445
, 2014, 12:25 a.m., Joshua Colp wrote:
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(Updated Sept. 14, 2014, 12:25 a.m
of the relay type.
Confirmed bidirectional media flow using the TURN relay.
Thanks,
Joshua Colp
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://reviewboard.asterisk.org/r/3999/#comment23811
Despite it not changing behavior I'd still use 20 here to match 12.
- Joshua Colp
On Sept. 16, 2014, 9:28 p.m., Jonathan Rose wrote:
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- Joshua Colp
On Sept. 15, 2014, 7:27
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Ship it!
Ship It!
- Joshua Colp
On Sept. 16, 2014, 9:28
://reviewboard.asterisk.org/r/4008/diff/
Testing
---
Modified the dialing API to change callerID at certain points (after call but
before handling responses, after handling responses). Confirmed that new code
correctly defers sending COLP updates.
Thanks,
Joshua Colp
=trunk, itsp, phone, etc. Lower level blocks remain
the same and additional logic on top can be added to handle this sort of
thing.
I look forward to seeing what others think!
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that people are familiar with they're fine.
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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.
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445 Jan Davis
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 but
implemented in
pjsip_configuration with endpoint
, identify, outbound registrations,
etc existing but that fact is an underlying detail they don't worry or
care about.
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George Joseph wrote:
On Sat, Sep 20, 2014 at 2:20 PM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
George Joseph wrote:
snip
I was thinking that we probably don't want to create hard coded
objects
called trunk, user, etc. Instead let the user
= notverysecret
context = trusted
disallow = all
allow = g722
mailboxes = 1000
Are you OK with a separate config file? It would make manipulating it
easier since there'd be no duplicate section names.
Yes.
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have an old install as well as a new?
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- Joshua Colp
On Sept. 19, 2014, 10:02
://reviewboard.asterisk.org/r/4016/#comment23871
This is actually incorrect. p-outboundproxy may point to a statically
allocated outbound proxy. To ensure both cases are met use ref_proxy to set it
to NULL. This will unref if it should and otherwise not unref.
- Joshua Colp
On Sept. 22, 2014, 8:30 p.m
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Ship it!
Ship It!
- Joshua Colp
On Sept. 23, 2014, 12:18
Paul Albrecht wrote:
On Sep 22, 2014, at 3:47 PM, Joshua Colp jc...@digium.com
mailto:jc...@digium.com wrote:
Paul Albrecht wrote:
Asterisk 13 beta2 compile fails:
.
.
.
[CC] chan_pjsip.c - chan_pjsip.o
[CC] pjsip/dialplan_functions.c - pjsip/dialplan_functions.o
[LD] chan_pjsip.o pjsip
handling responses). Confirmed that new code
correctly defers sending COLP updates.
Thanks,
Joshua Colp
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://reviewboard.asterisk.org/r/4032/#comment23885
What will this do for UDP transport? For those the transport should be
empty.
- Joshua Colp
On Sept. 29, 2014, 7:25 p.m., opticron wrote:
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On Sept. 29, 2014, 7:47 p.m., Joshua Colp wrote:
branches/12/res/res_pjsip_nat.c, line 49
https://reviewboard.asterisk.org/r/4032/diff/1/?file=67680#file67680line49
What will this do for UDP transport? For those the transport should be
empty.
opticron wrote:
udp is also
I'm down with proposal #2 but I have to ask: would this work with
realtime, and how?
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no candidates for RTCP?
It would still crash.
- Joshua Colp
On Oct. 1, 2014, 3:43 p.m., Kevin Harwell wrote:
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,
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On Oct. 7, 2014, 6:45 p.m., Joshua Colp wrote:
I don't think this goes far enough. What if I receive no candidates for
RTCP? It would still crash.
Kevin Harwell wrote:
I tested for this. The only time it crashes is when there are no RTP
candidates in the list
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- Joshua Colp
On Oct. 9, 2014, 3:44 p.m
---
Performed calls and ran tests.
Thanks,
Joshua Colp
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://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
(Encrypted)
Optional Optimistic On Success (Encrypted)
Thanks,
Joshua Colp
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/bridge_native_rtp.c 424850
Diff: https://reviewboard.asterisk.org/r/4057/diff/
Testing
---
Performed calls and ran tests.
Thanks,
Joshua Colp
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then the person had a valid license agreement at the time it
was uploaded.
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Check us out at: www.digium.com www.asterisk.org
/asterisk/res_pjsip.h 425395
Diff: https://reviewboard.asterisk.org/r/4073/diff/
Testing
---
Sent outgoing calls with various users (numbers, number+letters, letters) and
confirmed that user=phone was set when it should be.
Thanks,
Joshua Colp
/r/4073/#review13498
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On Oct. 13, 2014, 4:54 p.m., Joshua Colp wrote:
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Testing
---
Sent outgoing calls with various users (numbers, number+letters, letters) and
confirmed that user=phone was set when it should be.
Thanks,
Joshua Colp
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- Joshua Colp
On Oct. 13, 2014, 5:32 p.m., Matt Jordan wrote:
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(Updated
://reviewboard.asterisk.org/r/3997/#comment24038
No, AST_LIST_FIRST and AST_LIST_LAST would return the same thing since it's
a linked list containing one channel.
- Joshua Colp
On Oct. 13, 2014, 5:32 p.m., Matt Jordan wrote
the message went out.
Thanks,
Joshua Colp
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://reviewboard.asterisk.org/r/3997/#comment24060
glue1 will always exist so this can just become an else
- Joshua Colp
On Oct. 14, 2014, 7:11 p.m., Matt Jordan wrote:
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Otherwise g2g
- Joshua Colp
On Oct. 14, 2014, 7
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