Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
On Monday 15 June 2020 at 00:41:14, Bruce Ferrell wrote: > Way back in the mists of time, I built my asterisk installation with SNMP > support. Heh... I never even knew that was possible :) > That said, I actually prefer ARA/ARI to flat file configuration of endpoints > and dialplans. Changes are more or less instantaneous and easily shared > between instances. Agreed - ARA is a great system, and I really like that it can be combined with flatfile configs on a single server. > The ODBC way is a pain, so I tend to just use the native MySQL method Oh? What makes you say ODBC is a pain? I have two files (/etc/odbc.ini and /etc/idbcinst.ini, which are 8 lines and 3 lines in size respectively) and I had to install one file /usr/local/lib/libmaodbc.so to make it work with MariaDB. What's easier about the native MySQL method? > for ARA configuration as well as CDR collection. CDR reports are just a SQL > query away. Yes: a) efficient b) can be done on a machien remote from the call processing c) is realtime - call ends, CDRs are immediately available for analysis d) can even include triggers on DB updates, for example to raise anti-fraud alerts. Regards, Antony. -- Python is executable pseudocode. Perl is executable line noise. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
Way back in the mists of time, I built my asterisk installation with SNMP
support.
It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that
moment in time.
That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans. Changes are more or less instantaneous and easily shared between instances. The ODBC
way is a pain, so I tend to just use the native MySQL method for ARA configuration as well as CDR collection. CDR reports are just a SQL query away.
Only problem is all of the web ui's for Asterisk seem to be PHP that presumes
the flatfiles.
On 6/14/20 2:55 PM, Telium Technical Support wrote:
Just run ‘core show calls’ as a command from the AMI, and parse the results.
I don’t think there is an equivalent pure AMI command.
*From:*asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
Behalf Of *Jonathan H
*Sent:* Sunday, June 14, 2020 5:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show
calls"?
Wow! I've been *-ing for about 6 years and had literally no idea about that!
I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit
hacky) dialplan.
That's making me sound ungrateful. I don't mean to be!
On Sun, 14 Jun 2020, 22:39 Steve Edwards, mailto:asterisk@sedwards.com>> wrote:
On Sun, 14 Jun 2020, Jonathan H wrote:
> Thank you... but "just update the database" - hmm, what database?
I used MySQL.
> Did you mean ARI? I still can't find the command! The asterisk wiki is
> somewhat, um... spread around!
ARA as in Asterisk RealTime Architecture
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
https://www.voip-info.org/asterisk-realtime/
As I recall (back from 2015), you tell Asterisk which 'configuration file'
you want to read from MySQL like this:
# /etc/asterisk/extconfig.conf
[settings]
musiconhold.conf = mysql,vchat,static
; musiconhold.conf = mysql,vchat,musiconhold
I have no idea if this will help, but here are the tables as I defined them
back in 2015.
create table if not exists static
(
id int(11) not null auto_increment
, cat_metric int(11) not null default '0'
, var_metric int(11) not null default '0'
, commented int(11) not null default '0'
, filename varchar(128) not null default ''
, category varchar(128) not null default
'default'
, var_name varchar(128) not null default ''
, var_val varchar(128) not null default ''
, primary key (id)
)
;
-- defaults
set @CAT_METRIC = 0;
set @FILENAME = 'musiconhold.conf';
set @VAR_METRIC = 0;
-- Funk Dance
set @COMMENTED = 0;
set @NAME = 'Funk Dance';
insert into static set
cat_metric = @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric = @VAR_METRIC
, var_name = 'directory'
, var_val =
concat('/source/src/tmp/T2/moh/', @NAME, '/')
;
insert into static set
cat_metric = @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric = @VAR_METRIC
, var_name = 'mode'
, var_val = 'files'
;
insert into static set
cat_metric = @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric = @VAR_METRIC
, var_name = 'sort'
, var_val = 'random'
;
insert into static set
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
Just run ‘core show calls’ as a command from the AMI, and parse the results.
I don’t think there is an equivalent pure AMI command.
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Jonathan H
Sent: Sunday, June 14, 2020 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show
calls"?
Wow! I've been *-ing for about 6 years and had literally no idea about that!
I can see a way I could put it to a different use, but it seems to be a bit of
a sledgehammer to crack the walnut of "how many current callers" compared to
one line of (albeit hacky) dialplan.
That's making me sound ungrateful. I don't mean to be!
On Sun, 14 Jun 2020, 22:39 Steve Edwards, mailto:asterisk@sedwards.com> > wrote:
On Sun, 14 Jun 2020, Jonathan H wrote:
> Thank you... but "just update the database" - hmm, what database?
I used MySQL.
> Did you mean ARI? I still can't find the command! The asterisk wiki is
> somewhat, um... spread around!
ARA as in Asterisk RealTime Architecture
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
https://www.voip-info.org/asterisk-realtime/
As I recall (back from 2015), you tell Asterisk which 'configuration file'
you want to read from MySQL like this:
# /etc/asterisk/extconfig.conf
[settings]
musiconhold.conf= mysql,vchat,static
; musiconhold.conf= mysql,vchat,musiconhold
I have no idea if this will help, but here are the tables as I defined them
back in 2015.
create table if not exists static
(
idint(11) not null auto_increment
, cat_metricint(11) not null default '0'
, var_metricint(11) not null default '0'
, commented int(11) not null default '0'
, filename varchar(128) not null default ''
, category varchar(128) not null default 'default'
, var_name varchar(128) not null default ''
, var_val varchar(128) not null default ''
, primary key (id)
)
;
-- defaults
set @CAT_METRIC = 0;
set @FILENAME = 'musiconhold.conf';
set @VAR_METRIC = 0;
-- Funk Dance
set @COMMENTED = 0;
set @NAME = 'Funk Dance';
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'directory'
, var_val = concat('/source/src/tmp/T2/moh/',
@NAME, '/')
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'mode'
, var_val = 'files'
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'sort'
, var_val = 'random'
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'type'
, var_val = 'preset'
;
-- insert into static set
--cat_metric= @CAT_METRIC
-- , category = @NAME
-- , commented = @COMMENTED
-- , filename = @FILENAME
-- , var_metric= @VAR_METRIC
-- , var_name = 'application'
-- , var_val = '/usr/bin/mpg123 --mono -b 0 -f 8192
-q -r 8000 -s -@ http://206.190.136.141:5022/Live'
-- ;
-- FILES
-- set @COMMENTED = 0;
-- insert into static set
--cat_metric= @CAT_METR
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
Wow! I've been *-ing for about 6 years and had literally no idea about
that!
I can see a way I could put it to a different use, but it seems to be a bit
of a sledgehammer to crack the walnut of "how many current callers"
compared to one line of (albeit hacky) dialplan.
That's making me sound ungrateful. I don't mean to be!
On Sun, 14 Jun 2020, 22:39 Steve Edwards, wrote:
> On Sun, 14 Jun 2020, Jonathan H wrote:
>
> > Thank you... but "just update the database" - hmm, what database?
>
> I used MySQL.
>
> > Did you mean ARI? I still can't find the command! The asterisk wiki is
> > somewhat, um... spread around!
>
> ARA as in Asterisk RealTime Architecture
>
> https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
> https://www.voip-info.org/asterisk-realtime/
>
> As I recall (back from 2015), you tell Asterisk which 'configuration file'
> you want to read from MySQL like this:
>
> # /etc/asterisk/extconfig.conf
>
> [settings]
> musiconhold.conf= mysql,vchat,static
> ; musiconhold.conf= mysql,vchat,musiconhold
>
> I have no idea if this will help, but here are the tables as I defined
> them back in 2015.
>
> create table if not exists static
> (
>idint(11) not null auto_increment
> , cat_metricint(11) not null default '0'
> , var_metricint(11) not null default '0'
> , commented int(11) not null default '0'
> , filename varchar(128) not null default ''
> , category varchar(128) not null default
> 'default'
> , var_name varchar(128) not null default ''
> , var_val varchar(128) not null default ''
> , primary key (id)
> )
> ;
>
> -- defaults
> set @CAT_METRIC = 0;
> set @FILENAME = 'musiconhold.conf';
> set @VAR_METRIC = 0;
>
> -- Funk Dance
> set @COMMENTED = 0;
> set @NAME = 'Funk Dance';
> insert into static set
>cat_metric= @CAT_METRIC
> , category = @NAME
> , commented = @COMMENTED
> , filename = @FILENAME
> , var_metric= @VAR_METRIC
> , var_name = 'directory'
> , var_val =
> concat('/source/src/tmp/T2/moh/', @NAME, '/')
> ;
> insert into static set
>cat_metric= @CAT_METRIC
> , category = @NAME
> , commented = @COMMENTED
> , filename = @FILENAME
> , var_metric= @VAR_METRIC
> , var_name = 'mode'
> , var_val = 'files'
> ;
> insert into static set
>cat_metric= @CAT_METRIC
> , category = @NAME
> , commented = @COMMENTED
> , filename = @FILENAME
> , var_metric= @VAR_METRIC
> , var_name = 'sort'
> , var_val = 'random'
> ;
> insert into static set
>cat_metric= @CAT_METRIC
> , category = @NAME
> , commented = @COMMENTED
> , filename = @FILENAME
> , var_metric= @VAR_METRIC
> , var_name = 'type'
> , var_val = 'preset'
> ;
> -- insert into static set
> --cat_metric= @CAT_METRIC
> -- , category = @NAME
> -- , commented = @COMMENTED
> -- , filename = @FILENAME
> -- , var_metric= @VAR_METRIC
> -- , var_name = 'application'
> -- , var_val = '/usr/bin/mpg123 --mono -b 0 -f
> 8192 -q -r 8000 -s -@ http://206.190.136.141:5022/Live'
> -- ;
>
> -- FILES
> -- set @COMMENTED = 0;
> -- insert into static set
> --cat_metric= @CAT_METRIC
> -- , category = 'FILES'
> -- , commented = @COMMENTED
> -- , filename = @FILENAME
> -- , var_metric= @VAR_METRIC
> -- , var_name = 'directory'
> --
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
On Sun, 14 Jun 2020, Jonathan H wrote:
Thank you... but "just update the database" - hmm, what database?
I used MySQL.
Did you mean ARI? I still can't find the command! The asterisk wiki is
somewhat, um... spread around!
ARA as in Asterisk RealTime Architecture
https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
https://www.voip-info.org/asterisk-realtime/
As I recall (back from 2015), you tell Asterisk which 'configuration file'
you want to read from MySQL like this:
# /etc/asterisk/extconfig.conf
[settings]
musiconhold.conf= mysql,vchat,static
; musiconhold.conf= mysql,vchat,musiconhold
I have no idea if this will help, but here are the tables as I defined them
back in 2015.
create table if not exists static
(
idint(11) not null auto_increment
, cat_metricint(11) not null default '0'
, var_metricint(11) not null default '0'
, commented int(11) not null default '0'
, filename varchar(128) not null default ''
, category varchar(128) not null default 'default'
, var_name varchar(128) not null default ''
, var_val varchar(128) not null default ''
, primary key (id)
)
;
-- defaults
set @CAT_METRIC = 0;
set @FILENAME = 'musiconhold.conf';
set @VAR_METRIC = 0;
-- Funk Dance
set @COMMENTED = 0;
set @NAME = 'Funk Dance';
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'directory'
, var_val = concat('/source/src/tmp/T2/moh/',
@NAME, '/')
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'mode'
, var_val = 'files'
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'sort'
, var_val = 'random'
;
insert into static set
cat_metric= @CAT_METRIC
, category = @NAME
, commented = @COMMENTED
, filename = @FILENAME
, var_metric= @VAR_METRIC
, var_name = 'type'
, var_val = 'preset'
;
-- insert into static set
--cat_metric= @CAT_METRIC
-- , category = @NAME
-- , commented = @COMMENTED
-- , filename = @FILENAME
-- , var_metric= @VAR_METRIC
-- , var_name = 'application'
-- , var_val = '/usr/bin/mpg123 --mono -b 0 -f 8192
-q -r 8000 -s -@ http://206.190.136.141:5022/Live'
-- ;
-- FILES
-- set @COMMENTED = 0;
-- insert into static set
--cat_metric= @CAT_METRIC
-- , category = 'FILES'
-- , commented = @COMMENTED
-- , filename = @FILENAME
-- , var_metric= @VAR_METRIC
-- , var_name = 'directory'
-- , var_val = '/var/lib/asterisk/moh/'
-- ;
-- insert into static set
--cat_metric= @CAT_METRIC
-- , category = 'FILES'
-- , commented = @COMMENTED
-- , filename = @FILENAME
-- , var_metric= @VAR_METRIC
-- , var_name = 'mode'
-- , var_val = 'files'
-- ;
create table if not exists musiconhold
(
name varchar(80) not null
Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?
Thank you... but "just update the database" - hmm, what database? Did you mean ARI? I still can't find the command! The asterisk wiki is somewhat, um... spread around! On Sat, 13 Jun 2020 at 16:56, Steve Edwards wrote: > > On Sat, 13 Jun 2020, Jonathan H wrote: > > > I need to ensure that a MusicOnHold stream is only running when there's > > a caller on hold and listening.To do that, I need to rewrite and reload > > the moh.conf file when the caller hangs up IF there are no other callers > > (ie there's just 1 active call as the caller hangs up), and then > > rewrite and reload again when there's a new caller. > > How about ARA to configure MOH and then just update the database. > > -- > Thanks in advance, > - > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > https://www.linkedin.com/in/steve-edwards-4244281-- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > 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/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 17:05 schrieb Antony Stone: Hi Antony, > You mean that the Thomson phone is registering to Deutsche Telekom? > > I thought it was registering to your Asterisk server. Sorry, I didn't read correctly your test 2b... Normally my Thomson phone is registering to my Asterisk server. I tried to register the Thomson phone directly to Telekom's server, to check if the problem could be in my Asterisk... > Maybe it would be a good idea to tell *exactly* what your network setup is, > because I'd certainly assumed something that's clearly not true; maybe others > here have as well. Well, I'll try: - DSL-Modem, connected to a BananaPI with Debian 9 - On the BananaPI, PPPoE to connect to the Internet, iptables and some scripts to manage the Gateway and Firewall - Many VLANs, some of them can use the Internet via NAT - The phones are in an own VLAN without any routing to the Internet (exception for my phone was temporarily made to allow the tests) - In the phone's VLAN there is the Asterisk server, running on the same BananaPI the act as Gateway/Firewall - Mobile phone connected via WLAN in the same VLAN used from the PCs, and with routing to the Internet via NAT > Basically, what SIP phones (hardware or software) are you using, what are > they > registering to, and what role is Asterisk playing in all of this? How do > calls to/from the public phone network get routed from/to your telephones? We have two phones Thomson ST2022, registering to the Asterisk server. The Asterisk server registers to Deutsche Telekom and MessageNet. All calls are normally routed by Asterisk to Deutsche Telekom. Some *incoming* calls to an italian number arrives via MessageNet and will be directed to my Thomson phone. What I tried connecting the phone directly to the Internet and the servers of Deutsche Telekom was just a test, not the normal situation. Do I have explained my current situation? Of course I can send extract of configurations, if needed... What I'll do tomorrow with a test phone is: 1) connecting it to my Asterisk and try to make a call 2) connecting it directly to the servers of Deutsche Telekom (using my network) and try to make a call Thanks a lot for your help Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
On Sunday 14 June 2020 at 16:38:19, Luca Bertoncello wrote: > Am 13.06.2020 um 22:56 schrieb Antony Stone: > > Hi again, > > > 2b. Take your Thomson telephone to some other location with Internet > > access, let it register to your home Asterisk server, and then make a > > call to the same number yet again. > I tried it on the network of a friend. > Not possible to establish a connection at all... > I *suppose* Deutsche Telekom just allow a logon on their servers from > the IP of the user, Wait a moment... You mean that the Thomson phone is registering to Deutsche Telekom? I thought it was registering to your Asterisk server. Maybe it would be a good idea to tell *exactly* what your network setup is, because I'd certainly assumed something that's clearly not true; maybe others here have as well. Basically, what SIP phones (hardware or software) are you using, what are they registering to, and what role is Asterisk playing in all of this? How do calls to/from the public phone network get routed from/to your telephones? Antony. -- In science, one tries to tell people in such a way as to be understood by everyone something that no-one ever knew before. In poetry, it is the exact opposite. - Paul Dirac Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 16:48 schrieb Michael Keuter: Hi Michael, > the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß > PTMP with 3-10 numbers) is always tied to your DSL account. I supposed it... > There is a special "DeutschlandLAN SIP-Trunk Pure" where it does not depend > on your DSL account (as it is standard with most other VoIP providers). OK, I really don't think I want to subscribe this option just to check if the problem is in my account... :D Any other suggestion how to find *where* the problem is? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
> Am 14.06.2020 um 16:38 schrieb Luca Bertoncello : > > Am 13.06.2020 um 22:56 schrieb Antony Stone: > > Hi again, > >> 2b. Take your Thomson telephone to some other location with Internet access, >> let it register to your home Asterisk server, and them make a call to the >> same >> number yet again. I'm sure you can get the Thomson to connect to Asterisk >> via >> some external network, since you say you can do this from your Android >> phone. >> Again, check the call quality. > > I tried it on the network of a friend. > Not possible to establish a connection at all... > I *suppose* Deutsche Telekom just allow a logon on their servers from > the IP of the user, who tries to log on (with other words: my VoIP login > can just log on from my current IP)... Hi Luca, the standard Deutsche Telekom SIP-account (former ISDN Mehrgeräteanschluß PTMP with 3-10 numbers) is always tied to your DSL account. There is a special "DeutschlandLAN SIP-Trunk Pure" where it does not depend on your DSL account (as it is standard with most other VoIP providers). > This would explain why I didn't got my mobile phone connecting to the > Telekom's server and establish a call... > > I also tried to stop Asterisk and all other network services on my > Linux-Box Firewall/Gateway, including the traffic shaper (in the case, > this was the problem), then connect my Thomson phone to the Telekom's > server and call my father in law. > Always the same problem... > > So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290). > I'll connect it to my network and my Asterisk and will try to call my > father in law for a test. > > I really do *not* expect any change in the situation... I think, the > problem should be somewhere by Deutsche Telekom... > > What is your opinion? > > Btw: I did all tests with my father in law, since he had time for me > today, but the problem exists an almost all calls, incoming or outgoing, > no matter from/to which network provider... > > Thanks > Luca Bertoncello > (lucab...@lucabert.de) Michael http://www.mksolutions.info -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi again, > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the > same > number yet again. I'm sure you can get the Thomson to connect to Asterisk > via > some external network, since you say you can do this from your Android phone. > > Again, check the call quality. I tried it on the network of a friend. Not possible to establish a connection at all... I *suppose* Deutsche Telekom just allow a logon on their servers from the IP of the user, who tries to log on (with other words: my VoIP login can just log on from my current IP)... This would explain why I didn't got my mobile phone connecting to the Telekom's server and establish a call... I also tried to stop Asterisk and all other network services on my Linux-Box Firewall/Gateway, including the traffic shaper (in the case, this was the problem), then connect my Thomson phone to the Telekom's server and call my father in law. Always the same problem... So, tomorrow I'll get another VoIP phone from a colleque (Elmeg IP 290). I'll connect it to my network and my Asterisk and will try to call my father in law for a test. I really do *not* expect any change in the situation... I think, the problem should be somewhere by Deutsche Telekom... What is your opinion? Btw: I did all tests with my father in law, since he had time for me today, but the problem exists an almost all calls, incoming or outgoing, no matter from/to which network provider... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 14.06.2020 um 10:49 schrieb Antony Stone: >> So, the module voicemail in Asterisk does *not* have the same problems >> as the other phones. >> And the Thomson VoIP-phone has more problems than my Android connected >> to the Asterisk... > > So, you don't get "consistently good quality" in any situation, but it seems > like the Thomson phone being involved makes things worse. Correct >>> what happens if you let your Android phone connect via LTE to your home >>> Asterisk server and you dial your (home, cabled) Thomson phone from it? >>> What's the call quality like then? >> >> The quality is terrible. It is not possible to understand any word... >> BUT: if I call my wife using the Thomson (she uses a Thomsons, same >> model, too!) the quality is excellent... > > And is her Thomson connected on the same network to the same Asterisk server, > or is it somewhere else altogether? Yes, both telefons are in the same VLAN and Asterisk, too. > Why do you have: > >> allow=ilbc > > in sip.conf? I can't really remember why I added it... I try to remove it now and call my father in law. No changes... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
On Sunday 14 June 2020 at 09:00:26, Luca Bertoncello wrote: > I did another test, today: I called my leased line number using my > mobile phone (over GSM, not VoIP) and wait for the answering maschine. > So, as a normal call from outside if I'm not at home. > Result: the quality is *excellent*. I didn't hear any "interruptions" in > the message of the answering maschine and, as I played the message I > spoked there were no "interruptions", too... > > So, the module voicemail in Asterisk does *not* have the same problems > as the other phones. > And the Thomson VoIP-phone has more problems than my Android connected > to the Asterisk... So, you don't get "consistently good quality" in any situation, but it seems like the Thomson phone being involved makes things worse. > > what happens if you let your Android phone connect via LTE to your home > > Asterisk server and you dial your (home, cabled) Thomson phone from it? > > What's the call quality like then? > > The quality is terrible. It is not possible to understand any word... > BUT: if I call my wife using the Thomson (she uses a Thomsons, same > model, too!) the quality is excellent... And is her Thomson connected on the same network to the same Asterisk server, or is it somewhere else altogether? Why do you have: > allow=ilbc in sip.conf? Antony. -- The words "e pluribus unum" on the Great Seal of the United States are from a poem by Virgil entitled "Moretum", which is about cheese and garlic salad dressing. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
Re: [asterisk-users] Voice "broken" during calls
Am 13.06.2020 um 22:56 schrieb Antony Stone: Hi Antony, > I would like to see a much simpler one-for-one comparison: only change one > thing at a time, and see what the difference is. > > So: I suggest you try *two* independent *pairs* of tests: OK > 1a. Using your Android phone, connect using your home wireless network (I > assume you have a wireless network, if not then skip to test 2) to your home > Asterisk server, make a phone call to some external number, check the call > quality. Today was the quality not so excellent as yesterday... Both I and my father in law could hear "interruptions", but so so much as if I call with the Thomson phone... > 1b. Using your Thomson phone, connected using your home cabled network to > your > home Asterisk server, make a phone call to the same external number and check > the call quality. This was the same as always... More little "broken voice" on both parts... > 2a. Using your Android phone, connect from outside your home wireless network > over LTE to your home Asterisk server and make a phone call to the same > number > again (you'll need someone with a bit of patience and understanding on the > other end of this number ...) Check the call quality. It was a little bit better. I didn't hear any "interruptions", but my father in law does. Not many, but somes... > 2b. Take your Thomson telephone to some other location with Internet access, > let it register to your home Asterisk server, and them make a call to the > same > number yet again. I'm sure you can get the Thomson to connect to Asterisk > via > some external network, since you say you can do this from your Android phone. > > Again, check the call quality. Right now I don't have the possibility to do that... :( I did another test, today: I called my leased line number using my mobile phone (over GSM, not VoIP) and wait for the answering maschine. So, as a normal call from outside if I'm not at home. Result: the quality is *excellent*. I didn't hear any "interruptions" in the message of the answering maschine and, as I played the message I spoked there were no "interruptions", too... So, the module voicemail in Asterisk does *not* have the same problems as the other phones. And the Thomson VoIP-phone has more problems than my Android connected to the Asterisk... Maybe helps this behaviour to find the problem? > PS; Just for fun, what happens if you let your Android phone connect via LTE > to your home Asterisk server and you dial your (home, cabled) Thomson phone > from it? What's the call quality like then? The quality is terrible. It is not possible to understand any word... BUT: if I call my wife using the Thomson (she uses a Thomsons, same model, too!) the quality is excellent... > In regard to: > > On Saturday 13 June 2020 at 18:25:32, Luca Bertoncello wrote: > >> 2) where can I change these settings? > > sip.conf > > Look for lines such as > > disallow=all > allow=ulaw > allow=alaw > allow=h263 > > They may be in the [general] section, or they may be in the client (Android / > Thomson) specific sections. In my [general] section I have: disallow=all allow=alaw allow=ulaw allow=ilbc allow=g729 allow=g723 allow=gsm ; Messagenet need gsm... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ 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/asterisk-users
