Yes I use ast_waitfor with 2nd parameter -1 into a detached background
thread. The function ast_waitfor used into my application function
called from the dialplan and also into the dtmf read thread. This
behavior is observed after bridging the incoming channel(the channel
where I try to read
On July 30, 2014, 3:34 p.m., Mark Michelson wrote:
trunk/channels/chan_sip.c, lines 30871-30893
https://reviewboard.asterisk.org/r/3867/diff/1/?file=65719#file65719line30871
If you use ast_rtp_dtls_cfg_copy() above as I recommend, then this
entire block can be removed.
vassilux . wrote:
Yes I use ast_waitfor with 2nd parameter -1 into a detached
background thread. The function ast_waitfor used into my application
function called from the dialplan and also into the dtmf read thread.
This behavior is observed after bridging the incoming channel(the
channel where
Thank for the information about the frame hook.
I understand that both functions can be called by different threads because
lock/unlock used internally.
2014-08-04 11:44 GMT+02:00 Joshua Colp jc...@digium.com:
vassilux . wrote:
Yes I use ast_waitfor with 2nd parameter -1 into a detached
vassilux . wrote:
Thank for the information about the frame hook.
I understand that both functions can be called by different threads
because lock/unlock used internally.
If you are referring to ast_waitfor and ast_read those functions are NOT
safe to be called form separate threads
ok Joshua thank again
2014-08-04 12:28 GMT+02:00 Joshua Colp jc...@digium.com:
vassilux . wrote:
Thank for the information about the frame hook.
I understand that both functions can be called by different threads
because lock/unlock used internally.
If you are referring to ast_waitfor
I asked this on the users list a week and a half ago but haven't gotten any
response. I'm hoping someone here with PRI/ISDN experience can help guide me
in the right direction.
I have a dialplan (freepbx) that plays a busy signal in-band when an extension
is busy (before an Answer). Stripped
Run Progress before the playtones. This is documented in
https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application
--Eric
---
Frequently Asked Questions
Q1: How do a transfer a call using a Polycom phone?
A1: While on a call press the Transfer button on the phone,
Sorry for confusing the issue, I should have stripped out that line from the
dialplan as well. Given just the busy() line:
exten = 1005,n,Busy(20)
The busy tone should(?) be generated from the PRI channel driver. This is the
tone that the Telco is saying is being sent incorrectly. I've
Have you tried using Progress?
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Monday, August 04, 2014 11:58 AM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] How to diagnose early media on a PRI
Sorry
I have tried it, yes - the results are the same. When Busy() is called, the
channel driver gets a message and opens the early media stream if it hasn't
been opened already. I have the Q.931 entry for the alerting message for
Progress Description: Inband information or appropriate pattern now
Why do you want to use early media instead of using OOB signaling by using
Hangup(17)?
I'm not an expert on PRI, but maybe your telco is not passing on the early
media to the caller.
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Justin
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trunk/channels/chan_sip.c
I'm not an expert on PRI, but maybe your telco is not passing on the early
media to the caller.
Yes, I believe that to be the case as well. My Telco has been...less than
helpful. They are blaming the PBX, so I'm looking for a way to prove that
asterisk is doing things correctly and shift the
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trunk/channels/chan_sip.c
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There appears to be the potential for some refcounting badness
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(Updated Aug. 4, 2014, 2:42 p.m.)
Status
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This change has been
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(Updated Aug. 4, 2014, 3:25 p.m.)
Status
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This change has been
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This is missing a modification to tests/rest_api/tests.yaml.
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On Aug. 4, 2014, 4:10 p.m., Mark Michelson wrote:
/branches/12/contrib/ast-db-manage/voicemail/versions/39428242f7f5_increase_recording_column_size.py,
lines 36-37
https://reviewboard.asterisk.org/r/3870/diff/1/?file=65734#file65734line36
There is something just absolutely
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(Updated Aug. 4, 2014, 4:39 p.m.)
Review request for Asterisk Developers.
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Review request for Asterisk Developers.
Repository: Asterisk
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Review request for Asterisk Developers.
Bugs: ASTERISK-24150
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/trunk/channels/iax2/include/codec_pref.h
I'm not an expert on PRI, but maybe your telco is not passing on the early
media to the caller.
Yes, I believe that to be the case as well. My Telco has been...less than
helpful. They are blaming the PBX, so I'm looking for a way to prove that
asterisk is doing things correctly and
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