So this turned out more complicated than I originally thought!
My expectation:
Verbosity gets logged using an "at least" check against the current
system's verbose level, which if passed subsequently gets checked against
the logging channel's verbose level. Thus only verbose messages with a
On Fri, Sep 10, 2021 at 12:44 PM Jerry Geis wrote:
> HI All,
>
> I am trying to get SIPml5 working with 18.6.0.
> My http.conf file:
> enabled=yes
> bindaddr=myip
> bindport=8088
> serverName=MyName
> tlsenabled=true
> tlsbindaddr=myip
> tlscertfile=/etc/letsencrypt/live/mpname/fullchain.pem
>
>
d on your system so the
backtrace doesn't have any extractable information. Please see the wiki [3]
on how to get a useful backtrace.
Before that though I recommend upgrading to the latest version of Asterisk
[1]. Or if you're set on using a certified version [3]. The version you are
on is quite old,
res_format_attr_g729.c -> res_format_attr_g729.o
>
>
> Is this to be expected or should I make a bug report?
>
>
When you pulled the lasted code this change would have forced a
re-configure. If you haven't already try doing a full clean and rebuild,
and see if you still have the error:
$ mak
flag is still needed in Asterisk. However, the setting
appears to propagate in some way into the bundled pjproject configuration.
This is in some way affecting the build of pjproject, then subsequently
causing res_pjsip in Asterisk to not load at runtime. Further investigation
is required as to why th
e --with-ssl is
> used? I could not find a clear explanation for this problem and how to fix
> it
>
There appears to be a bug here. I configured, built, and ran with the same
options mentioned (--with-ssl, etc...) and received similar pjsip module
load errors
splay/AST/Asterisk+16+Function_CHANNEL
--
Kevin Harwell
Senior Software Developer
Sangoma Technologies
Check us out at: https://sangoma.com & https://asterisk.org
___
asterisk-app-dev mailing list
asterisk-app-...@lists.digium.com
http://lists.d
;
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo
nguage = en
>
> internal_sample_rate = 0
>
> mixing_interval = 20
>
> record_file_append = no
>
> max_members = 10
>
> video_mode = follow_talker
>
>
>
> [4]
>
> type = user
>
> admin = no
>
> marked = no
>
> startmuted = no
>
> mus
So you are probably seeing it work or not in Chrome vs Firefox due to
browser, and codec support of such occurrences.
--
Kevin Harwell
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: https://digium.com & https:
;
>
>
> Is the wiki web page mistaken or is this an actual http.conf setting that
> is undocumented?
>
The page is mistaken. It should not be there. the 'tlscafile' option is not
supported by the Asterisk http server. I've removed it from the wiki.
Thanks for catching that!
>
>
>
ded branchc and [1] but met no success yet
>
> Best regards
>
> [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml
>
>
--
Kevin Harwell
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - U
lpful too if you have those. Which channel type (chan_sip,
local channel, chan_pjsip) is involved, and how you are enabling the jitter
buffer (dialplan function vs configuration) would be good to know as well.
[1] https://issues.asterisk.org
[2] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Ba
nt and res_pjsip_multihomed was removed as
the bulk of its code was moved into the res_pjsip core.
--
Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digiu
s did not go out with the latest release of
13.7.0. Actually the new StatsD Dialplan application currently resides in
master only. A small change to the res_statsd api was made and got tagged
with that issue number for some reason, thus making it look as if the
StatsD application feature was added to 13
/display/AST/Configuring+res_pjsip
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip
[3]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide
Hope that helps,
--
Kevin Harwell
Digium, Inc. | Software Developer
445
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com wrote:
07.03.2015 0:24, Kevin Harwell пишет:
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com
wrote:
Hello.
Asterisk 13.2.
I transfer configs from chan_sip to res_pjsip.
In chan_sip i have
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks,
--
Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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