Hi Jaco,
I do know what to look for with deadlocks, and the odd thing is that
what I pasted in the earlier post is not your run of the mill mutex
based deadlock. Four locks are currently held and something inside each
thread is just not finishing. None of the threads are waiting for
On 10/30/15 16:04, Mark Murawski wrote:
I'm getting the following lockup very randomly in 11.20. This is pretty
devastating to one of my production systems.
===
=== 11.20.0
=== Currently Held Locks
Howdy All,
I'm working on trying to output the channel related to an rtp stream. I
found this line in ast_rtp_read()/res_rtp_asterisk.c and this pointed me
to a starting position:
ast_verb(4, "%p -- Probation passed - setting RTP source address to
%s\n", rtp, ast_sockaddr_stringify())
I
I'm getting the following lockup very randomly in 11.20. This is pretty
devastating to one of my production systems.
===
=== 11.20.0
=== Currently Held Locks
On 11/16/15 10:59, Matthew Jordan wrote:
Things to investigate:
(1) Get a gdb backtrace when this occurs, so you can find out exactly
what the AstDB synchronization thread is doing that is blocking for a
long period of time.
(2) Find out *exactly* what is occurring on your system. If the
On 11/17/15 15:39, Mark Murawski wrote:
On 11/16/15 10:59, Matthew Jordan wrote:
SNIP
Either way, nothing above makes me think there is a bug in Asterisk.
Here's a completely different box, different arch/hardware/kernel but
compiled from the same exact asterisk source.
I hit
Hey,
So, I'm seeing an issue where a Polycom IP-550 with 4.1.1 firmware is
sending RFC2833 DTMF packets as shown in the capture attached. I can
send pcaps as necessary, if needed.
10.0.90.6 is the Polycom 10.0.90.1 is Asterisk.
So of course you do:
> rtp set debug 10.0.90.6
And then get
into Read()
[2017-04-04 17:21:41.557] [C-175d] Got RTP RFC2833 from
10.0.90.6:8110 (type 101, seq 016608, ts 3618850425, len 04, mark 1,
event 0003, end 0, duration 00160)
On 04/04/2017 04:52 PM, Mark Murawski wrote:
Er, small correction. Asterisk clearly shows RTP
Er, small correction. Asterisk clearly shows RTP *flowing*, but not
receiving DTMF from 10.0.90.6
On 04/04/17 16:37, Mark Murawski wrote:
Hey,
So, I'm seeing an issue where a Polycom IP-550 with 4.1.1 firmware is
sending RFC2833 DTMF packets as shown in the capture attached. I can
send
On 4/6/17 4:23 PM, Matt Fredrickson wrote:
Hey Mark,
First off, thanks for reaching out to the Asterisk community to talk
about the trouble you're having :-)
This actually is a development related mailing list - so primarily for
development of Asterisk C source code level discussions,
Hi!
I have the following situation here:
;;;
; WAN1 and traffic to PBX-A / PBX-B
[transport-udp]
type = transport
symmetric_transport= yes
protocol = udp
bind =
Testing. 3, 2, 1.
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Thanks!
On 2/27/21 2:53 PM, Joshua C. Colp wrote:
On Sat, Feb 27, 2021 at 1:57 PM Mark Murawski
mailto:markm-li...@intellasoft.net>> wrote:
H!,
I'm working on a cross-channel standard log line for processing new
calls. I had no issue implementing this for DAHDI/IAX/CH
H!,
I'm working on a cross-channel standard log line for processing new
calls. I had no issue implementing this for DAHDI/IAX/CHAN_SIP, but I
have this so far for pjsip got this far, but this does not seem like the
right approach because occasionally reads to the sip dialogue are
reading
onf
Response: Error
EventList: start
Message: Errors/Warnings will follow
etc etc
On 1/5/22 18:06, Mark Murawski wrote:
Hi!
Throwing my .02 in here. Adjust to .10 for inflation!
On 1/4/22 14:53, aster...@phreaknet.org wrote:
However, there are a lot of dialplan problems that repres
Hi!
Throwing my .02 in here. Adjust to .10 for inflation!
On 1/4/22 14:53, aster...@phreaknet.org wrote:
However, there are a lot of dialplan problems that represent potentially
valid syntax that will cause an error at runtime, such as branching to
somewhere that doesn't exist. The
Hi Michal,
I would recommend you loop up with other users/developers who still need
chan_sip and start a new repo for it. It probably won't be officially
endorsed by Sangoma/Asterisk but it's definitely possible to keep it
alive with the right team.
On 10/29/22 12:40, Michal Rybarik
Howdy -dev,
Looking for a starting point to investigate this issue:
Using the following AEL (and Running straight-dialplan results in the
same issue)
context services {
2802 => {
Answer();
Set(GLOBAL(foo)=${CHANNEL});
ConfBridge(Agent-PJSIP/2802);
}
9000 => {
Answer();
On 1/15/24 04:42, Joshua C. Colp wrote:
On Sun, Jan 14, 2024 at 9:10 PM Mark Murawski
wrote:
... snip...
Goal:
Extract out an individual speaker's audio that's going into a ConfBridge
(Cannot join the bridge as a member, because then you're getting all
participants
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