On Thu, 7 Nov 2019 at 15:24, Joshua C. Colp wrote:
> On Thu, Nov 7, 2019 at 11:14 AM Steve Davies wrote:
>
>> Hi,
>>
>> So I've chosen to post this in -dev in case someone manages to come up
>> with a pointer to an existing patch, because if not I imagine I
Hi,
So I've chosen to post this in -dev in case someone manages to come up with
a pointer to an existing patch, because if not I imagine I will do the work
and submit it myself. Advice, corrections and opinions are most welcome.
I noticed that 'res_monitor' is flagged as deprecated on the basis
Hi,
This is possibly an OLD bug that has existed since at least 1.8.x and
persists into 16.x.
In asterisk 16, main/translate.c ~line 646 is the following code:
if (f->samples != current->samples && ast_test_flag(current,
AST_FRFLAG_HAS_TIMING_INFO)) {
ast_debug(4, "Sample size different
Amazing, clear answer. Thank you so much Richard. That documentation link
is also very helpful.
Regards,
Steve
On Mon, 20 Nov 2017 at 17:56 Richard Mudgett <rmudg...@digium.com> wrote:
> On Mon, Nov 20, 2017 at 7:03 AM, Steve Davies <davies...@gmail.com> wrote:
>
>> Hi,
Hi,
Perhaps the answer to this will be pointing me at some documentation - That
is fine, but I've failed to find it so far, so forgive me if the following
is a dumb question.
With Asterisk-11 I added a built-in feature which allowed a very customised
version of stopping call recording on a
Hi,
I believe that the current state of affairs with Asterisk's SIP over TCP or
TLS registration is that if a connection is dropped or closed, then the
registration is allowed to persist.
Given that a re-connect will almost certainly not be from the same IP/port
pair, should a TCP or TLS
If chan_sip really has to go, then perhaps the place to start would be a
marketing effort?
* Why is PJSIP better?
* What does PJSIP do that chan_sip does not?
* Is there anything in chan_sip that PJSIP does not do, and how do you
solve that?
...
This can then be followed with bribery
* What
A classic issue which bit me a few weeks ago was that the permissions on
the core dump directory changed, preventing asterisk from creating a file?
Also, check the contents of the kernel sysctl that specifies the filename
of core dumps - If it contains an absolute path, it will override
;jc...@digium.com> wrote:
> Steve Davies wrote:
> > Not sure how/where else to report this...
>
> Issues should always be reported on the issue tracker[1] and to that
> extent I've gone ahead and created one for this[2] since it's a
> regression. If there's additional inf
Not sure how/where else to report this...
commit 5e6b1476a087407a052f007d326c504cfeefebe7
ASTERISK-25614
2 code paths which approximate the following will cause a lock-inversion
deadlock:
approximate call orders are:
a)
pj_timer_heap_poll (PJ_LOCK)
ast_rtp_on_ice_complete
Hi,
I believe there is a known issue in Postgres where if multiple threads
call pgsql_exec(), then it can cause threads to get stuck inside Postgres -
This is why I suggested that Postgres patch, which puts Asterisk locks
around the call to pgsql_exec().
OTOH, I don't think you mentioned what DB
Hi,
I have a randomly occurring de-registration of SIP devices since 11.20.0,
and I believe that the new scheduler is indirectly the cause. The following
assumes that the new scheduler can (and will reasonably regularly) re-use
the sched-id of 0, which never used to be possible.
Scenario:
-
I also experienced a lockup using realtime postgres. I this what you are
doing? ou do not describe your environment in much detail.
If so, have you seen this patch which is in GIT for asterisk 11?
diff --git a/res/res_config_pgsql.c b/res/res_config_pgsql.c
index be2d4ed..bce7394 100644
---
I finally found the wiki.asterisk.org pages that I needed to read to get
started with git and gerrit as configured in these parts. For reference to
help anyone else trying to migrate from the SVN/reviewboard system:
https://wiki.asterisk.org/wiki/display/AST/Git+Usage
Hi,
I submitted that change, as we have need for not the 'no' and the 'never'
cases on different devices/trunks etc, and before the patch they were
almost the same.
I completely agree with the above suggestion from Kevin. I always set
progressinband manually for all of my device definitions, so
I've been pulled up on almost every reviewboard entry I raised recently for
missing some piece of information. In each case, the relevant field has
indeed been missing, and I have felt like a bit of an idiot.
Today I discovered the cause... At least for the Linux Chrome browser.
If a field in a
Thanks.
I had cherry-picked the patch from Reviewboard # 3624 and probably missed
the symbols bit somehow.
Steve
On 27 June 2014 17:00, Joshua Colp jc...@digium.com wrote:
Steve Davies wrote:
Hi,
Kia ora,
I do run a heavily patched version, but is the export of the symbol
Hi,
I captured a segfault (sadly without a core) on a customer site as follows:
asterisk[16862]: segfault at 770 ip 08098fa0 sp b483bdc0 error 4 in
asterisk[8048000+18f000]
# addr2line -e asterisk 08098fa0
/usr/src/asterisk-1.8.26.0/main/cdr.c:741
740:for (; cdr; cdr = cdr-next) {
741:
Hi,
I've been looking at ASTERISK-22079, and what might have caused it. The
comment I added to the bottom of that ticket does not explain the
segfault/backtrace that is attached to that ticket, so I have kept digging.
Fundamentally, I would like to know what stops a scheduled event from being
Hi,
Thanks for looking at this.
On 29 January 2014 16:34, Matthew Jordan mjor...@digium.com wrote:
[snip]
You'll note that the refcall is only called if ast_sched_del
(eventually) returns a valid ID that it deleted.
Agreed. In my proposed failure example, ast_sched_del is called first,
and
I believe there is a regression in the CDR code since 1.8.11. It also
affects versions 10 and 11, but probably not version 12.
https://issues.asterisk.org/jira/browse/ASTERISK-22954
After a SIP attended transfer, the code that was added in
ASTERISK-16990 tries to copy userfield etc CDR data onto
Hi,
Could someone let me know who is the maintainer of Astmanproxy (Dave @
Popvox seems to have vanished)
I have a growing patch, and a long line of people asking me for it -
I'd like to be able to pass it over the fence to an official
maintainer.
Thanks,
Steve
Hi,
Firstly, a note of caution for those compiling MMX into Zaptel... I
use an EPIA processor, which may be part of the problem, but enabling
the Zaptel MMX code caused sufficient corruption that it affected
other userspace processes! I suddenly started getting page-errors in a
web application,
Hi,
We have an asterisk user who has wireless phones - I believe that
these phones are causing occasional locked-up SIP channels. While this
is a problem in itself, it is not easily diagnosable, so I am looking
for a solution from a different angle.
The problem occurs when a ZAP channel hangs
On 5/13/05, Steve Davies [EMAIL PROTECTED] wrote:
[snip]
The failure was due to an inability to authenticate the SUBSCRIBE, and
this was happening when the phone was trying to SUBSCRIBE using
credentials from SIP/line1, and find_peer(...) was returning
credentials for the correct phone
Hi,
A couple of days ago I posted regarding SIP 'SUBSCRIBE's failing from
our snom190 phones here:
http://lists.digium.com/pipermail/asterisk-users/2005-May/106545.html
I did some more analysis, and it seems that the problem is caused
because each of our phones REGISTERs as two or more users
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