from:"Floimair Florian"

[asterisk-users] Question regarding SIP 183 Session Progress handling in Asterisk

2009-06-30 Thread Floimair Florian

Dear Asterisk community!

 

I am having trouble with a project concerning the 183 Session Progress SIP 
messages. Asterisk seems to only accept these when there is also a Session 
Description (SDP) included in the message.

I also verified this by looking at the code.

 

However for a project we are working with a trunk to a third party system 
(Alcatel) and they are insisting that this behavior is non-compliant with 
RFC3261 (SIP). So can someone please tell me the reason,

why Asterisk does not support 183 messages without SDP as this would really 
help me finding arguments in this situation. So far Alcatel just tells us that 
this is not SIP-compliant and that we have to change things

on the Asterisk side, but I'm not quite sure that this is really the case and 
having arguments could help me clarifying this situation.

 

Thanks in advance.

 

With best regards

Florian Floimair
Technical Support

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25 312
Sip: f.floim...@commend.com 
file:///T:\KAT\Signaturen\=%22sip:f.floim...@commend.com%22 
Fax: +43-662-85 62 26
f.floim...@commend.com mailto:f.floim...@commend.com 
http://www.commend.com http://www.commend.com/ 

Security and Communication by Commend

FN 178618z | LG Salzburg

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] pcapsipdump or general sip debug question - the solution

2017-01-17 Thread Floimair Florian

Or you may use sngrep if you prefer command line tools

 
 
With best regards


-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Yves
Gesendet: Dienstag, 17. Januar 2017 14:20
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] pcapsipdump or general sip debug question - the 
solution

Hi,

i know about this feature and use it a lot...
my question was, how to get pcapsipdebug to generate only one file...

BUT... meanwhile I found out how to accomplish this easy task.

1.) open first pcap file in wireshark
2.) open second pcap file in wireshark using the menu "file -> merge"
3.) go to "telephony -> sip flows"
4.) select the two "legs" of the call
5.) klick button "flow sequence" et voilà... one ladder diagram exactly the way 
I needed it

thanks anyways,
yves

Am 17.01.2017 um 12:34 schrieb Jean Aunis:
> Hello,
>
> There is a built-in tool in Wireshark for this : menu Telephony => 
> Voip Calls, the select your call and click on "Flow Sequence".
>
> Best regards
>
> Jean Aunis
>
>
> Le 17/01/2017 à 12:27, Yves a écrit :
>> Hi,
>>
>> I am using pcapsipdump for debugging sip calls.
>>
>> when I have to debug a call, pcapsipdump generates two files per 
>> call... one for the sip dialog between the client (softphone) and the 
>> server (asterisk) and one for the sip dialog between the server 
>> (asterisk) and the sip registrar... is there a way to get this into 
>> one file ? the objective is to see both sides of the call in a single 
>> ladder diagram or just to have more comfort in analyzing the full 
>> flow within wireshark.
>>
>> If this is not possible, is there a free tool for sip (together with
>> rtp) debugging that is able to catch the full sip flow between both 
>> ends of one call in a single file (per call) with pcap compatibility 
>> (including the rtp packets)?
>>
>> thank you
>> yves
>>
>>
>
>


--
_
-- 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] Asterisk realtime - Error with index length in alembic script

2017-07-12 Thread Floimair Florian

Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)

 
 
With best regards

Florian Floimair 


-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance

 
 
With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
https://linkprotect.cudasvc.com/url?a=http://www.commend.com=E,1,AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,=1

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: 
https://linkprotect.cudasvc.com/url?a=https://asterisk-users-boun...@lists.digium.com=E,1,7s7_D_Myc9BrXsqexg-b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jqolYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,=1
 
https://linkprotect.cudasvc.com/url?a=https://[mailto:asterisk-users-boun...@lists.digium.com=E,1,b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,=1]
 Im Auftrag von Thomas
Gesendet: Montag, 10. Juli 2017 14:07
An: 
https://linkprotect.cudasvc.com/url?a=https://asterisk-users@lists.digium.com=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW-0IaAjeofhcEGlqHiUa9FX1v_0Z61fjn6Cglc1LwJESdZ5CsnB1ZeUMPn7gV2z5agkz3kh8onHV0Oxnmn9c4DjH6CBU=1
Betreff: [asterisk-users] ConfBridge increase talking volume as standard

Hello,

is it possible to increase talking volume for caller in ConfBridge as standard 
without need to press buttons after joining an conference room.

best regards
Thomas 

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYVFcBMK4FJNKC_QOHqI9aSjt=1

-- 
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,VVFR1vt-VTtY8TkmHZUWfvj_qWyvKuVj3j3kEWWRkDTqIoK5793fkmzz9AnT3OcPsNe5K5B3f4dXsXnP4GaT1rCJ3ejq9IG69CQTwZrDqsi13OegC5RQ8hdx=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,8XODZp3VpPc6qb0tcihfIZdGrdIrkd-VCre0cSW0BegTWE_FQwvkwDTHGm8U3zoh7SeoujNXlQu50jrPRfD-eWh-imQTu3o3wFA6ag-rUcunf1TZRY11=1

-- 
_
-- Bandwidth and Colocation Provided by http://www

[asterisk-users] Asterisk realtime - Error with index length in alembic script

2017-07-12 Thread Floimair Florian

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
e96a0b8071c_increase_pjsip_column_size.py
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance

 
 
With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Thomas
Gesendet: Montag, 10. Juli 2017 14:07
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] ConfBridge increase talking volume as standard

Hello,

is it possible to increase talking volume for caller in ConfBridge as standard 
without need to press buttons after joining an conference room.

best regards
Thomas 

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,3cxgVpYz6rj8HJh87TiGg9vmNONVb8R9gj8CUtsKQo4J7XZd3A8P5Q2lgkuqRb7I2h0ILWV9fb2VVtM_fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,8wdFEBPawJhe3Q7z9CDJspktcWZmak6_F7Qwy-KlgT8Y3RKfi8rz6GZboEEYt3CZnt4-JjH7gKYVY79x72M6dUv0yXmSnCduYVFcBMK4FJNKC_QOHqI9aSjt=1

-- 
_
-- 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

[asterisk-users] Asterisk realtime in combination with ARI - error while trying to prepare SQL statement for writing into database

2017-07-13 Thread Floimair Florian

Hey guys!

I successfully got Asterisk realtime (14.6.0) with MariaDB (MySQL fork) running 
on Debian 9.

I will document the steps to do so shortly (the main difference is default 
encoding and the odbc connector & its configuration).

What I’m trying to do now is to use ARI to create PJSIP endpoints as outlined 
in this wiki article:
https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration

So far so good. Excecuting the examples using curl PUT aor and auth was working 
fine, when I ran the line for the endpoint object I got an error in the 
Asterisk CLI and log stating:

res_config_odbc.c: SQL Prepare failed! [INSERT INTO ps_endpoints (id, 
rpid_immediate, device_state_busy_at, t38_udptl_maxdatagram, dtls_rekey, 
named_pickup_group, direct_media_method, send_rpid, pickup_group, sdp_session, 
dtls_verify, message_context, mailboxes, record_on_feature, dtls_private_key, 
dtls_fingerprint, from_domain, timers_sess_expires, named_call_group, 
dtls_cipher, media_encryption_optimistic, aors, identify_by, callerid_privacy, 
mwi_subscribe_replaces_unsolicited, cos_audio, context, rtp_symmetric, 
transport, moh_suggest, t38_udptl, fax_detect, tos_video, srtp_tag_32, 
use_avpf, call_group, fax_detect_timeout, sdp_owner, force_rport, callerid_tag, 
rtp_timeout_hold, use_ptime, media_address, voicemail_extension, rtp_timeout, 
set_var, contact_acl, force_avp, record_off_feature, from_user, send_diversion, 
t38_udptl_ipv6, tone_zone, language, allow_subscribe, rtp_ipv6, callerid, 
moh_passthrough, cos_video, asymmetric_rtp_codec, ice_support, aggregate_mwi, 
one_touch_recording, mwi_from_user, accountcode, allow, rewrite_contact, 
user_eq_phone, rtp_engine, subscribe_context, auth, 
direct_media_glare_mitigation, trust_id_inbound, bind_rtp_to_media_address, 
disable_direct_media_on_nat, media_encryption, media_use_received_transport, 
allow_overlap, dtmf_mode, outbound_auth, tos_audio, dtls_cert_file, 
dtls_ca_path, dtls_setup, connected_line_method, g726_non_standard, 100rel, 
timers, direct_media, acl, timers_min_se, trust_id_outbound, sub_min_expiry, 
rtcp_mux, send_pai, rtp_keepalive, t38_udptl_ec, t38_udptl_nat, allow_transfer, 
dtls_ca_file, outbound_proxy, inband_progress) VALUES (?, ?, ?, ?, ?, ?, ?, ?, 
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?,
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, 
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, 
?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, ?, 
?, ?, ?, ?, ?)]

Carefully comparing the attributes from the JSON output of the curl statement 
(which was successfully completed) and the columns in the database I found the 
culprit:

The database is missing the following attribute: „dtls_fingerprint“
I prepared my database using the alembic scripts from the 14.6.0 source. After 
manually adding the column to the database and restarting Asterisk the 
statement was successfully executed and everything works fine now.

In addition to the missing attribute there are few columns in the database 
table that do not match to a JSON attribute in the pjsip endpoint object.
These are:

-  contact_deny

-  contact_permit

-  contact_user

-  deny

-  disallow

-  external_media_address

-  permit

-  redirect_method


So far to my findings.

Should I file a bug-report for this?


With best regards

Florian Floimair


-- 
_
-- 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] Asterisk realtime - Error with index length in alembic script

2017-07-13 Thread Floimair Florian

Yeah, I will do that.

I have some more things I need to clarify, so I will just gather all the 
information and sum it up beforehand.



With best regards

Florian Floimair


Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Marcelo Terres
Gesendet: Mittwoch, 12. Juli 2017 22:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Please open a Ticket (https://issues.asterisk.org), to let them know that they 
need to update the documentation in Wiki and also handle this situation when 
using Alembic in Debian 9 (could happens in other Distros too).

Marcelo H. Terres <mhter...@gmail.com<mailto:mhter...@gmail.com>>
IM: 
mhter...@jabber.mundoopensource.com.br<mailto:mhter...@jabber.mundoopensource.com.br>
https://www.mundoopensource.com.br<https://linkprotect.cudasvc.com/url?a=https://www.mundoopensource.com.br=E,1,cC98P-aUo0sGW0nm4x61IEJa3J0afB1zbPdps3H6n3PIgLTX7AqK3Tu23_zNx520Yyc8Yzs47UV9i25C0XsWCC24S8fQAXstahwvJHpIxRcmrMu0_FU,=1>
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 12 July 2017 at 13:11, Floimair Florian 
<f.floim...@commend.com<mailto:f.floim...@commend.com>> wrote:
Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/<https://linkprotect.cudasvc.com/url?a=https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/=E,1,vcstDda1EXiLuEfYH_bWSEQiKUNSlxsJREZBgYMwpFTlFa1RWIuFe-eoxVvQIGuxWq4LaHyhrIGhulqolz16NPWIGtPYLbSURFxq9b0Tl5B4r3vd56NqBNAzGA,,=1>).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)



With best regards

Florian Floimair


-Ursprüngliche Nachricht-
Von: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>>
Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance



With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
https://linkprotect.cudasvc.com/url?a=http://www.commend.com=E,1,AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,=1

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: 
https://linkprotect.cudasvc.com/url?a=https://asterisk-users-boun...@lists.digium.com=E,1,7s7_D_Myc9BrXsqexg-b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jqolYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,=1
 
https://linkprotect.cudasvc.com/url?a=https://[mailto:asterisk-users-boun...@lists.digium.com=E,1,b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,=1]<https://linkprotect.cudasvc.com/url?a=https://%5bmailto:asterisk-users-boun...@lists.digium.com=E,1,b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,=1%5d>
 Im Auftrag von Thomas
Gesendet: Montag, 10. Juli 2017 14:07
An: 
https://linkprotect.cudasvc.com/url?a=https://asterisk-users@lists.digium.com=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW-0IaAjeofhcEGlqHiUa9FX1v_0Z61f

Re: [asterisk-users] PJSIP list of peers online/offline?

2017-06-29 Thread Floimair Florian

You can try:

pjsip show endpoints

However, there is no summary line in the end (only the total number of objects)
so you will have to parse the status of each entry yourself to get these
statistics.

With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Antony Stone
Gesendet: Mittwoch, 28. Juni 2017 12:59
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] PJSIP list of peers online/offline?

Hi.

I have some Nagios / Icinga monitoring plugins I've created for Asterisk, and
one of them checks the percentage of SIP accounts which are currently
registered on an Asterisk server.

It does this by running "sip show peers" via AMI and analysing the summary line
at the end:

1066 sip peers [Monitored: 747 online, 310 offline Unmonitored: 3 online, 6
offline]

I then calculate 747 divided by (747+310) and report the % online (because I
know I'm not interested in the unmonitored ones).

However, a customer has upgraded one of their servers from Asterisk 11 to
Asterisk 13, and "sip show peers" no longer works.

I can see a whole list of commands starting with "pjsip" but there's no "pjsip
show peers", so what's the new command which will tell me how many online and
how many offline SIP peers there are?

Thanks in advance,

Antony.

--
Never write it in Perl if you can do it in Awk.
Never do it in Awk if sed can handle it.
Never use sed when tr can do the job.
Never invoke tr when cat is sufficient.
Avoid using cat whenever possible.

Please reply to the list;
please *don't* CC me.

--
_
-- Bandwidth and Colocation Provided by
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,qLqu-5o6figiWLM--g8os87LkUGeDNP10U1PZ9AcHCdxpPfolVS1TIpyvjje_lGsVSylPENMEZOlqFJcfpfJ9i5N3qhrkw5_qnRyWOEBeFrVvQ,,=1
--

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:

https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,hnO0US2fleV36DGE9cMn7pFY0mbQF-GmabZg5DJbSPB7f5jH_zIySFyDBwL75JrqJwifVHjHZOjKtgB9weh50LUUUeI4CNAZmAkk4uAeBdxJIms,=1

--
_
-- 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] Asterisk realtime - Error with index length in alembic script

2017-08-01 Thread Floimair Florian

For anyone interested I documented my findings as a comment to the original 
wiki article here:

https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime?refresh=1501592246325=1501592309852=1501592385401=37455051#comment-37455051



With best regards

Florian Floimair


Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian
Gesendet: Donnerstag, 13. Juli 2017 11:52
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Yeah, I will do that.

I have some more things I need to clarify, so I will just gather all the 
information and sum it up beforehand.



With best regards

Florian Floimair

Von: 
asterisk-users-boun...@lists.digium.com<https://linkprotect.cudasvc.com/url?a=https://asterisk-users-boun...@lists.digium.com=E,1,0XqWlrx_jzejrGNUHtTyaJK4qUM_42T294RO02AeZx1yC7S4VAhPghx0V3eABtc6iCyi2Hw94tXfogxQe7-_d4nHtaVKyDRhuV9ENoTvLdcyt5Oj=1>
 
[mailto:asterisk-users-boun...@lists.digium.com<https://linkprotect.cudasvc.com/url?a=https://%5bmailto:asterisk-users-boun...@lists.digium.com=E,1,c178FD0VgmrFePlnP2rVqBbNajHEfk_KMZsBP5ySVdU8rGasTlFhG89zoM3W1egVUCBtu8PcUeBsekqYzsXH9pDuQBZo1kfs4ZeMdbh9=1>]
 Im Auftrag von Marcelo Terres
Gesendet: Mittwoch, 12. Juli 2017 22:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com<https://linkprotect.cudasvc.com/url?a=https://%26lt;asterisk-users@lists.digium.com=E,1,Dm7JnZlS9SB0f1GpjBHWKwoW8bbEMCNe9IfvHNMjrm3moyvQ-k1T4SHRcAh35JxByJanXoMJU-ydogOJWdWr9lWzPV6UzSVJqyTbgkkfRkQs0CZdzykzYw,,=1>>
Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Please open a Ticket (https://issues.asterisk.org), to let them know that they 
need to update the documentation in Wiki and also handle this situation when 
using Alembic in Debian 9 (could happens in other Distros too).

Marcelo H. Terres <mhter...@gmail.com<mailto:mhter...@gmail.com>>
IM: 
mhter...@jabber.mundoopensource.com.br<mailto:mhter...@jabber.mundoopensource.com.br>
https://www.mundoopensource.com.br<https://linkprotect.cudasvc.com/url?a=https://www.mundoopensource.com.br=E,1,cC98P-aUo0sGW0nm4x61IEJa3J0afB1zbPdps3H6n3PIgLTX7AqK3Tu23_zNx520Yyc8Yzs47UV9i25C0XsWCC24S8fQAXstahwvJHpIxRcmrMu0_FU,=1>
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 12 July 2017 at 13:11, Floimair Florian 
<f.floim...@commend.com<mailto:f.floim...@commend.com>> wrote:
Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/<https://linkprotect.cudasvc.com/url?a=https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/=E,1,vcstDda1EXiLuEfYH_bWSEQiKUNSlxsJREZBgYMwpFTlFa1RWIuFe-eoxVvQIGuxWq4LaHyhrIGhulqolz16NPWIGtPYLbSURFxq9b0Tl5B4r3vd56NqBNAzGA,,=1>).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)



With best regards

Florian Floimair


-Ursprüngliche Nachricht-
Von: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com<mailto:asterisk-users@lists.digium.com>>
Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for

Re: [asterisk-users] taskprocessor.c: The 'sorcery/contact-00000015' task processor queue reached 1500 scheduled tasks.

2017-11-24 Thread Floimair Florian

I have seen this log message as well. No clue yet as to what the reason for
this might be.
Any hint is appreciated.

With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Brian Capouch
Gesendet: Donnerstag, 23. November 2017 23:26
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] taskprocessor.c: The 'sorcery/contact-0015' task
processor queue reached 1500 scheduled tasks.

Running 15.1.2. I have four devices transitioned to use pjsip.

After about 1-2 days of uptime, psjip stops accepting registrations, and the
messages log contains the entry as per the subject.

At any given time, "pjsip show contacts" only shows the four devices.

Could someone point me to a fix, short of rebooting the server every day?

Thanks.

--
_
-- Bandwidth and Colocation Provided by
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,BXUoGx7Dyi10ofQjmmu-dM9Yf3o67rlYye3nwB50xPOsFPdXQD6UEXmxkuUZwjniYYuYjXk6Q9dN1YPr7kpAXBX-9is4q4t6PIyg-Cuu61Q,=1
--

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:

https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,2UYIus7YM1tHfSMRmGtMClG0tr14HGzed8tcbXmkxVLULQIQT_2ZMidTNFSB8NJivHh8XCahCIZ-trCXvq_6e08WgQrSlu_cGoPCfq0g6IlxcmNiIEI,=1

--
_
-- 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] [OT] Overview of Homer installation on Debian Stretch

2017-12-15 Thread Floimair Florian

@1) Not on the host, as jessie is only used within the container but you may 
run it on a Stretch host of course.

@2) The HOMER API contains modules for mysql and postgresql. The Debian 
maintainers simply split the resulting packages into three subpackages 
(homer-api general parts, db parts for mysql, and db parts for postgresql). So 
if you plan on using mysql you do not have to install the postgresql package. 
The methodology is the same as for asterisk modules in Debian’s asterisk 
packages. You can simply choose to only install the parts you really need.



With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg


Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Olivier
Gesendet: Freitag, 15. Dezember 2017 12:05
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: Re: [asterisk-users] [OT] Overview of Homer installation on Debian 
Stretch

Thanks Florian for replying.
1. I thought using Homer docker option required a Jessie setup.
Is it correct ?
2. Do you understand what homer-api-mysql package is for ?
After reading package's description on Debian stretch repo, it still keeps 
mystery to me.
Cheers

-- 
_
-- 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] [OT] Overview of Homer installation on Debian Stretch

2017-12-14 Thread Floimair Florian

  1.  The simplest option would be to run the Docker multicontainer-setup as 
described on the HOMER wiki here: 
https://github.com/sipcapture/homer/wiki/Docker-Install
  2.  Sure it is. Just edit your homer configuration to reflect the remote DB 
server
  3.  If you really want to learn and understand what’s happening, try 
following a manual setup from source as described here: 
https://github.com/sipcapture/homer/wiki/Quick-Install#-manual-setup-from-source-advanced

I have recently setup a Debian Stretch Azure VM along with an Azure MySQL 
database to run based on 3. There were a few hurdles to overcome (that for the 
most part were DB related) but I’ve managed to get there. Doing it this way 
helped me understand how things interact and where to look at in case of 
problems. It may take some time (as the wiki is not always very clear on every 
step) but it’s time well spent and worth investing.

If you really get stuck somewhere I might be able to help you.



With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Olivier
Gesendet: Dienstag, 12. Dezember 2017 16:59
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [asterisk-users] [OT] Overview of Homer installation on Debian Stretch

Hello,
I've discovered homer-api-postgresql and homer-api-mysql packages in Stretch 
repo.
I'm not sure I understand how Homer-API relates to Homer.
My questions are:
1. What is the simplest available installation option to install Homer on a 
dedicated box, this dedicated box gathering data from one or several Asterisk 
systems on the same LAN ?
2. Is it possible to centralize data on a remote Postgres/MySQL server (remote 
meaning hee Homer server acting as a DB client)

3. Suggestions ?
Cheers


-- 
_
-- 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

[asterisk-users] Asterisk Realtime PJSIP - slow output on "pjsip show xxxxx" commands

2018-06-18 Thread Floimair Florian

Hi all!

I have an Asterisk instance (15.4.1) with a MySQL DB in realtime configuration 
(using MySQL ODBC connector).

I noticed, that when I issue “pjsip show endpoints” in the CLI this takes 
forever (on average about 1 minute with 1029 entries in ps_endpoints table).
If I query the database over the same ODBC connector with isql, this takes less 
than a second.

I have enabled caching in sorcery.conf but it doesn’t really help (at least not 
concerning the output of “pjsip show x” commands).

Any idea what might be causing this and how I can optimize it?



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

From: asterisk-users  on behalf of 
David P 
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

Date: Monday, 18. June 2018 at 04:26
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Only 8kHz recorded after disallowing all but G722 
codec on inbound

I also just tried adding this:

 same => n,Set(SIP_CODEC_INBOUND=g722)

On Sat, Jun 16, 2018 at 4:36 PM David P 
mailto:davidswalkab...@gmail.com>> wrote:
We want to record inbound channels at 16kHz, but send only 8kHz to our peers. 
I've set our default profile in sip.conf to disallow all but g722, and the 
peers disallow all but ulaw. We have a proxy in front of Asterisk that is 
configured to disallow all but G722 also.

My test calls show inbound to the proxy is recorded at 16kHz, inbound in 
Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not 
working is Asterisk's sampling rate on inbound, and it seems to be downsampling.

After a lot of web searching, I can't find any explanation of why we're not 
getting 16kHz for G722. We're using Asterisk 14.7.6.

Cheers,
David
-- 
_
-- 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] Big leap - 1.8 to 15.4.0

2018-05-28 Thread Floimair Florian

You more or less have to follow the guidelines as if you were doing one step at 
a time (in terms of version) upgrades.

So you should consider the following documents in the source tree:

UPGRADE-10.txt
UPGRADE-11.txt
UPGRADE-12.txt
UPGRADE-13.txt
UPGRADE-14.txt
UPGRADE.txt (which would be 15)

Asterisk 15 still has chan_sip as well so I would give that a try before also 
doing the switch to chan_pjsip. Otherwise your upgrade task will be a lot 
bigger.
 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 28.05.18, 08:36 schrieb "asterisk-users im Auftrag von Stefan Viljoen" 
:

Hi all

We have to upgrade soon from prehistoric Asterisk 1.8.32.0 to Asterisk 
15.x.x (whatever minversion is current at the time.)

We are quite heavily invested into 1.8.32.0 at about 17 sites locally and 
internationally and have a LOT of custom software running our sites via AMI 
from various applications.

What are big gotchas to watch out for? Especially regarding PJSip as I've 
been following posts and forums and it seems theres massive changes there, 
nevermind just overall changes in jumping years of dev ahead on the Asterisk 
version...

What should I watch out for so that our sites don't collapse completely on 
telecoms functionality when we go to 15 from 1.8? Any resources I can refer to? 
No upgrade guide I can find cover -this- wide a leap in versions.

I assume virtually every site's dialplan will have to be rewritten from 
scratch, which in itself is already a massive undertaking.

Anybody with some advice?

Thanks,

Stefan


-- 
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.api-digital.com=E,1,KRg-WGCOt09FKuiTbY2Vvk9VdI1XyTlf3CFQZTZygsOfEW4CSC_dT0tlk6Evz1OdJsixcctGnZP2QACYB0cJDkab2EndnxhqXG457arg2g,,=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.digium.com%2fmailman%2flistinfo%2fasterisk-users=E,1,Yv6xd_siPyTM5153wuPRRE7Wx2W1npVL6c0P_v510iZ6BEuNP9A93rWg3i8-rkSVIjJuqswrbPIqXOIsb2OXNbKfv7eojfY2655ipMzI6w0lL-zlrt1t77HkXuc,=1

-- 
_
-- 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

[asterisk-users] Logging ARI debug messages

2018-01-11 Thread Floimair Florian

Hi there!

Is there any way I can turn on debug for ARI and sending the output to a 
separate log file?
So far I have only been able to turn on ARI debugging in the console which 
results in the debug output being logged in /var/log/asterisk/messages

I would love to have ARI debug log messages in /var/log/asterisk/debug or even 
better in it's own ari-debug file.



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-- 
_
-- 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] Logging ARI debug messages

2018-01-11 Thread Floimair Florian

Thanks for the quick reply Joshua!

I might dig into this and try an implementation.

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
Gesendet: Donnerstag, 11. Januar 2018 16:35
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Logging ARI debug messages

On Thu, Jan 11, 2018, at 11:30 AM, Floimair Florian wrote:
> Hi there!
> 
> Is there any way I can turn on debug for ARI and sending the output to 
> a separate log file?
> So far I have only been able to turn on ARI debugging in the console 
> which results in the debug output being logged in /var/log/asterisk/ 
> messages
> 
> I would love to have ARI debug log messages in /var/log/asterisk/debug 
> or even better in it's own ari-debug file.

That is not something anyone has implemented as of this time. The messages 
themselves just get raised as normal verbose messages.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
https://linkprotect.cudasvc.com/url?a=https://www.digium.com=E,1,GYHMzrUkSEpC6ZF8Ihfu0XRmq2T-03NlWDnRyWqARteU_a7mRZrKYUqXVNCn_gJ8_CR-6x5pNQmHfv-HpNi7bp7k00ykQwJm4H_7WUlpy0qv2DMF=1
 & www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,8BlAQ4nDh-7Z-3zxc8V5GJ1BXCYpjZGs21Kv62IKgyGb0uEg79Y_gbtu75F2JinT1i7a19PybUwpjfz0Ty9zeCSWZciuNppr2jTQsD8VrdUFjgni=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,eYyTp9Dxprjv-X1EtteMmKKbtPmiTgbHI_YcNWuBuRHPIb7RikqO95r6xHa03atqSDXZ2Um7nP5DN0LmOK6IpoLksgoRxIOKBA3ZCKNg=1

-- 
_
-- 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] asterisk mysql contacts

2018-01-17 Thread Floimair Florian

Yes there is.

You can follow the ODBC section in this document:

https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Paul Neuwirth
Gesendet: Mittwoch, 17. Januar 2018 18:42
An: John Kiniston 
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: Re: [asterisk-users] asterisk mysql contacts

On Wed, 17 Jan 2018 09:26:28 -0700
John Kiniston  wrote:

> use func_odbc, create a new function that does a lookup.
> 
> [CALLERID]
> prefix=LOOKUP
> dsn=MyDB
> readsql=SELECT CALLERID from MyNames where CallerIdNum = 
> '${SQL_ESC(${ARG1})}'
> 
> exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${CALLERID(NUM)}))
> 
> 
> On Wed, Jan 17, 2018 at 6:16 AM, Atux Atux  wrote:
> 
> > Hi. i have an asterisk 11 installation that i run in my soho 
> > environment. My system has mysql to store all the cdrs.
> > I would like make use of the mysql and store numbers and names. eg
> > +4922123456789 "Atux Null". So when the +4922123456789 calls in my
> > system the name "Atux Null" will pop up next to the number.
> > at the moment i have a database called MyNames in mysql that has 
> > this information, but i do not know how to make the dialplan read 
> > from this database.
> > I would like to ask if there is a way to implement this easily in my 
> > dialplan, please.
> >
> >

that is a good approach.. thinking over implementing this too.. and filling the 
numbers using a trigger on cdr table.. 
is there a good manual how to set up odbc connection to mysql?  I actually 
never used odbc on non microscrap systems..

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,wvMG11IAUjITKI9-y3nyVs3P0RZJsq30__4xDz-72btjUUH7bIQSCS6WS2L7TfK4uNbT5HR1_ddQxdmzXQozIG-BkpakxNG__r6VfbMmrnn_MAxMYZHm-uXP=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,Nw3s7x8mQWnEoAnyQapORcAM3-ksUAhgqQPgOIJ3dG1arLzQ3jW5MDlVqbrYQe_M5Km5UatyfRI1AY0WEq1l8Q6l6goT61cis4RiKxhCGUiPT8CLyPwBlf-gc3ZN=1

-- 
_
-- 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] What does pct mean?

2018-02-12 Thread Floimair Florian

No you're reading it wrong.

There are 188K received with no loss, and 16441K transmitted.

Still 8809 does not sound like a percentage to me  so there is something wrong 
with either the label or the value.
From what's in the code, you can see it's clearly a lost Packet count not a 
percentage.
So I guess Pct in this case is short for "Packet".

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Maier
Gesendet: Montag, 12. Februar 2018 17:46
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] What does pct mean?

Hi Carsten,

On 02/11/2018 at 07:46 PM Carsten Bock wrote:
> Hi,
> 
> Lost percent (%)

Are you sure? I'm seeing here:

...Receive. .Transmit..
CountLost Pct  Jitter   CountLost PctJitter  RTT
188K  00   0.000188K   16641K 8809   0.000   0.026

=> This doesn't sound reliable to me: there are 188K packets and 16641K of them 
are lost?! The Pct value is fluctuating between about 6009 and 9009.

Thanks,
Michael


> 
> 
> Am 11.02.2018 19:27 schrieb "Michael Maier" :
> 
>> Hello,
>>
>> could somebody please tell me the meaning of "Pct" as seen in asterisk cli:
>>
>> ...Receive. .Transmit..
>> CountLost Pct  Jitter   CountLost Pct  Jitter RTT
>>
>>
>> Thanks,
>> Michael

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http://www.api-digital.com=E,1,uxGzivqW6oZ241hxc5A3oz1rZf6JLog-Gi4ziwN-95NzbE_HEndRkD8LLLem_gvmmxd5k_T95J2jepit1IpIkZ2AxkG4RSoADT-AMulX4hxaaQ,,=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http://lists.digium.com/mailman/listinfo/asterisk-users=E,1,wIdvPI63-ImQqWZFblrmlGJbQ_cbmu31TlSPHYv9kAkHrNRLdIAjL2IUr9sVYxm6piHc0Pf2Zna7zuNOJ2hb4CVzp2WYVDsgZJqAyHt2YkVuGQ,,=1
-- 
_
-- 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

[asterisk-users] Encrypting passwords in config files

2018-08-16 Thread Floimair Florian

Hey there!



I was wondering what the best practice is concerning passwords in Asterisk's 
config files.



ari.conf has a neat feature where one can use a pre-encrypted password by using

password_format=crypt

for an ARI user



However, I was wondering how to do similar things with e.g. database 
credentials when using realtime.

Right now I am using a plain-text password in res_odbc.conf to get the database 
connection working.

So the only protection here is restricting file permissions of the config file.



Two questions that arise from this:



Is there any other way to do this that I am missing?

If no, would it be a desirable to implement pre-encrypted passwords for other 
config files in the same way as it is done in ari.conf?









 

 

With best regards



Florian Floimair

Innovation - Software-Development



COMMEND INTERNATIONAL GMBH

A-5020 Salzburg, Saalachstraße 51

http://www.commend.com 



Security and Communication by Commend



FN 178618z | LG Salzburg


-- 
_
-- 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] Autoreply ( Autoreply (Re: getting invites to rtp ports ??))

2018-09-10 Thread Floimair Florian

Can an administrator please throw out i...@online4you.nl of the mailing list. 
It's a bit annoying when one third of the messages is always the one below.

Thanks!

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 10.09.18, 00:01 schrieb "asterisk-users im Auftrag von i...@online4you.nl" 
:

Bedankt voor uw bericht.

Online4You is sinds 1 augustus niet meer operationeel. Per e-mail hebben 
wij u geinformeerd over de omstandigheden en uw opties.

Helaas kunnen wij u niet meer helpen, uw mail wordt niet doorgestuurd en/of 
beantwoord.

Indien uw abonnement is overgenomen door KovoKs, kijk dan voor 
contactgegevens op https://www.kovoks.nl/.

Dank voor uw vertrouwen de afgelopen jaren!

Met vriendelijke groet,

Online4You B.V.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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 --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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] Asterisk behind NAT Early Media Video

2018-04-11 Thread Floimair Florian

I did a quick check between what I have set and your settings below.

You can try the following and see if it helps

In your endpoint:
bind_rtp_to_media_address=yes




With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Benjamin Marty
Gesendet: Mittwoch, 11. April 2018 08:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 
<asterisk-users@lists.digium.com>
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

I think I found the root cause. The H264 Early Media video is received 
successfully on the Asterisk Server. It also seems to get processed. But it's 
send to the private IP of the receipent SIP phone.
For clarification:
178.82.XX.XX is my Public IP of my Internet access. Both phones use this as 
Public IP via standard Source NAT.
159.89.XX.XX is the IP of the Asterisk Server. For this test I used a Server 
without Destination NAT. So the eth0 interface has this IP.
Packet capture:
No. Time  SourceDestination 
  Protocol Length Info
141 2018-04-11 06:40:03.306561178.82.XX.XX  159.89.XX.XX
H264 64 PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408 SPS

Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst: da:81:42:3d:d0:e7 
(da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
User Datagram Protocol, Src Port: 5006, Dst Port: 13182
Real-Time Transport Protocol
H.264

No. Time  SourceDestination 
  Protocol Length Info
142 2018-04-11 06:40:03.306682159.89.XX.XX192.168.XX.XX 
H264 64 PT=H264, SSRC=0x5EE97C55, Seq=30572, Time=319121408 SPS

Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst: IETF-VRRP-VRID_6e 
(00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 10298, Dst Port: 5022
Real-Time Transport Protocol
H.264
PJSIP.conf:
[7004]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
;force_rport = yes
disallow = all
allow = g722, alaw, ulaw, gsm, ilbc, h264
aors = 7004
auth = auth7004

[7004]
type = aor
max_contacts = 2

[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004
extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})


2018-04-10 16:43 GMT+02:00 Benjamin Marty 
<benjamin.ma...@gmail.com<mailto:benjamin.ma...@gmail.com>>:
I just noticed, the calling device isn't even sending the early media video 
stream. It just sends an early media audio stream. Is there propably a change 
in the signaling needed?
(On another P2P SIP Server the early media video works.)

2018-04-10 12:29 GMT+02:00 Benjamin Marty 
<benjamin.ma...@gmail.com<mailto:benjamin.ma...@gmail.com>>:
Hi Florian
I already have the external_media_address set in the PJSIP setup. Also the 
external_signaling_address is set to the Public IP. If I make a call from an 
Early Media (video) capable device to an Early Media capable device (also 
video) the Early Media audio works perfectly. But no video. If I sniff 
with wireshark on the recipent device I just see G711 (audio) RTP traffic. The 
h264 RTP traffic is missing before I accept the call. After accepting the call 
the h264 RTP traffic comes through.
The 183 SIP protocoll comes through. Even Asterisk is noticing it:
-- PJSIP/6002-0013 is making progress passing it to PJSIP/6001-0012

I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13 with 
sip.conf (chan_sip). In both cases I just put the both case AST_FRAME_VIDEO: 
statements before the two voice cases, like in your diff and 
recompiled/reinstalled.
Regards
Benjamin


2018-04-10 9:37 GMT+02:00 Floimair Florian 
<f.floim...@commend.com<mailto:f.floim...@commend.com>>:
Hi Benjamin!

You're obviously using a similar scenario that I have in place for testing.
I initially had issues with early media (not only video also audio) as well in 
that scenario. What I had to do was to additionally set

external_media_address=

in pjsip.conf

Also, as I wrote the patch for early-media video I'd be interested in any 
feedback from it.




With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com<https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.commend.

Re: [asterisk-users] Asterisk behind NAT Early Media Video

2018-04-10 Thread Floimair Florian

Hi Benjamin!

You're obviously using a similar scenario that I have in place for testing.
I initially had issues with early media (not only video also audio) as well in 
that scenario. What I had to do was to additionally set

external_media_address=

in pjsip.conf

Also, as I wrote the patch for early-media video I'd be interested in any 
feedback from it.


 
 
With best regards

Florian Floimair
Innovation - Software-Development -  VoIP & DevOps

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
Gesendet: Montag, 9. April 2018 18:15
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video

On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
> wohoo, so if I unterstand it correctly with that patch early media 
> video works over the Asterisk server? In other words the Asterisk 
> server get's able to (process/)forward the early media video stream with that 
> patch?

The patch forwards video while in an early media state before the call is 
answered and bridged, yes.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: 
https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.digium.com=E,1,fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnEpbgeqpqYbk39h-m_lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduNnaBqVCDk,=1
 & www.asterisk.org

--
_
-- Bandwidth and Colocation Provided by 
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.api-digital.com=E,1,XToemLgPy6NQVyb_dF1q0qXSk-3YylF6rmIrWQvPhspxagnF5G63VHCU2nB67YHjZewMQU1rUCME4JBQMFPmNOCpc6ESOin_3Al6kti-lRo,=1
 --

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:
   
https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.digium.com%2fmailman%2flistinfo%2fasterisk-users=E,1,6VfJH-ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmol-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc=1

-- 
_
-- 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] Early or Pre VIdeo

2018-02-26 Thread Floimair Florian

Hi John!

There is no "clean" way yet that I am aware of, however I got it running by 
modifying the code in the Dial() application.
By default Dial() blocks anything other than audio in the early-media stage (no 
idea if this is wanted behavior or just not yet implemented).

I have a patch in the works that I will submit in the next few days.


With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
Tel: +43-662-85 62 25
Fax: +43-662-85 62 26
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von John T. Bittner
Gesendet: Sonntag, 25. Februar 2018 01:31
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [asterisk-users] Early or Pre VIdeo

Does anyone know if asterisk 15 supports Video before answering a call.

I am running PJSIP and have tested a bunch of settings, progress, answer call 
before calling the other phone and  even set the devices to send direct media.
Source video device is a door phone that is supposed to support early video.

Video works ok once answered.

Any help on this would be very helpful and appreciated.

Thanks

John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax:   201.806.2604
Cell:   973.390.1090
www.xaccel.net

CONFIDENTIALITY NOTICE:
This e-mail message, including any attachments, is for the sole use of the 
intended recipient(s) and may contain confidential
and privileged information which should not be shared or forwarded. Any 
unauthorized review, use, disclosure or distribution
is prohibited. If you are not the intended recipient, please contact the sender 
by reply e-mail and destroy all copies of the e-mail.

-- 
_
-- 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

[asterisk-users] chan_pjsip: DTMF mode "auto_info" on endpoints

2018-09-26 Thread Floimair Florian

Hey all!

I recently tried the dtmf_mode "auto_info" on my setup to support endpoints 
that only understand SIP INFO as a fallback.

My setup is the following:

Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO)

Both are configured with "auto_info" dtmf_mode in pjsip.conf. 
What I ran into is, that DTMF sent from endpoint A to endpoint B is 
additionally sent via inband audio on the RTP stream from Asterisk to endpoint 
B, as one can clearly hear the DTMF tone in the audio stream, when a capture is 
played back on Wireshark. On the leg from endpoint A to Asterisk there is no 
inband DTMF signal in the RTP audio stream.

Can someone confirm this behavior? If yes than this is clearly a bug.
I had a look in the code which introduced this feature and couldn't find 
anything obvious why this is happening.

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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] chan_pjsip: DTMF mode "auto_info" on endpoints

2018-09-27 Thread Floimair Florian

Yeah, I did enable core debug level 5 but I don't get something that would 
point me in the right direction.

["2018-09-26 11:39:28.6507"] DTMF[105612][C-0001] channel.c: DTMF end '1' 
received on PJSIP/X-0001, duration 250 ms
["2018-09-26 11:39:28.6507"] DTMF[105612][C-0001] channel.c: DTMF begin 
emulation of '1' with duration 250 queued on PJSIP/X-0001
["2018-09-26 11:39:28.9137"] DTMF[105612][C-0001] channel.c: DTMF end 
emulation of '1' queued on PJSIP/X-0001
["2018-09-26 11:39:28.9139"] DEBUG[105597][C-0001] chan_pjsip.c: Told to 
send end of digit on Auto-Info channel PJSIP/X- RFC4733 NOT 
negotiated using INFO instead.
 ["2018-09-26 11:39:28.9139"] DEBUG[105391] res_pjsip_session.c: Method is INFO

Here's my plan now:
The chan_pjsip_digit_begin() function in channels/chan_pjsip.c does not have 
any debug output in the code, only chan_pjsip_digit_end() does and I do see 
that in the log.
Maybe I need to insert additional debug statements in chan_pjsip_digit_begin() 
and see if it's really doing the right stuff in the case AST_SIP_DTMF_AUTO_INFO.
Another step would probably be then to have a look into 
ast_rtp_instance_dtmf_begin().

By the way I forgot to mention that Asterisk actually does send SIP INFO with 
DTMF to Endpoint B (which is what is expected) but it seems that the inband 
audio is added additionally (which is a problem).

With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com <http://www.commend.com/>

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 26.09.18, 15:47 schrieb "asterisk-users im Auftrag von Joshua Colp" 
:

On Wed, Sep 26, 2018, at 10:25 AM, Floimair Florian wrote:
> Hey all!
> 
> I recently tried the dtmf_mode "auto_info" on my setup to support 
> endpoints that only understand SIP INFO as a fallback.
> 
> My setup is the following:
> 
> Endpoint A (RFC4733) --> Asterisk <-- Endpoint B (SIP INFO)
> 
> Both are configured with "auto_info" dtmf_mode in pjsip.conf. 
> What I ran into is, that DTMF sent from endpoint A to endpoint B is 
> additionally sent via inband audio on the RTP stream from Asterisk to 
> endpoint B, as one can clearly hear the DTMF tone in the audio stream, 
> when a capture is played back on Wireshark. On the leg from endpoint A 
> to Asterisk there is no inband DTMF signal in the RTP audio stream.
> 
> Can someone confirm this behavior? If yes than this is clearly a bug.
> I had a look in the code which introduced this feature and couldn't find 
> anything obvious why this is happening.

Have you bumped up the core debug to see what's going on underneath? There 
will be information about whether it is really generating the DTMF in the core, 
and if so then it'd be a result of the digit_begin function of chan_pjsip 
returning a value it shouldn't.

-- 
Joshua Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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 --

Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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] how to use a database

2018-12-10 Thread Floimair Florian

Alembic currently doesn't cover queue_logs.
As of now it only covers configuration, voicemail and cdr.

With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 07.12.18, 15:56 schrieb "asterisk-users im Auftrag von hw" 
:

On 12/07/2018 03:36 PM, Administrator TOOTAI wrote:
> Le 07/12/2018 à 14:32, hw a écrit :
> 
> [...]
>>
>> Queues seem to be the only way to have several phones ring at once, or 
>> are there other ways?
> 
> Dial(SIP/Phone1/Phone2&.../Phonex,,)
> 

Good to know, thanks!

What are the entries needed in the queue_members table when using odbc? 
Alembic made the primary key so that each queue can only have one entry 
(What is an interface here?), and there's probably a reason for that. 
How do you enter several members for a queue?  Asterisk seems to either 
rather crash than to create a queue, or to do nothing.

-- 
_
-- 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] Detecting a fax

2019-01-11 Thread Floimair Florian

I would guess from your explanation that the "outgoing" call somehow ends up in 
your Asterisk machine again, either at the voicetest or fax extension.
You don't answer it in either of the extensions. 
That's what TOOTAI meant.

If this is done in another extension, than this part of the Dialplan is missing 
in your post. So we can only guess what's going on.
Either post the full dialplan or add some log or cli output (with verbose set 
to at least 1) so we can have a chance of seeing what's going on.

With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 11.01.19, 10:23 schrieb "asterisk-users im Auftrag von Neil Youngman" 
:

On 11/01/2019 09:19, Administrator TOOTAI wrote:
> Le 11/01/2019 à 10:12, Neil Youngman a écrit :
>> A while back, I posted about detecting when a call was picked up by a 
>> fax machine. It was suggested that having a "fax" extension and 
>> "faxdetect=yes" would cause it to jump to the "fax" extension. This 
>> was not something I could get to work.
>>
>> I have now created a very simple example. In sip.conf I have 
>> "faxdetect = yes". My example extension is:
>>
>> [test]
>> ;
>> ; Voice test extension
>> ;
>> exten => voicetest,1,NoOp()
>>  same => n,LOG(Notice,${CHANNEL}: Extension voiceout starting)
>>  same => n,LOG(Notice,${CHANNEL}: Starting Answer Machine Detection)
>>  same => n,AMD()
>>  same => n,LOG(Notice,${CHANNEL}: Answer Machine Detection 
>> ${AMDSTATUS}/${AMDCAUSE})
>>  same => n,Playback(/var/lib/asterisk/sounds/en/demo-congrats)
>>  same => n,LOG(Notice,${CHANNEL}: Voice out extension complete)
>>  same => n(hangup),Hangup()
>>
>>
>> ;
>> ; Fax detected extension
>> ;
>> exten => fax,1,NoOp()
>>  same => n,LOG(Notice,${CHANNEL}: Extension fax starting)
>>  same => n,LOG(Notice,${CHANNEL}: Fax Machine Detected)
>>  same => n,Playback(/var/lib/asterisk/sounds/en/silence/2)
>>  same => n,LOG(Notice,${CHANNEL}: Fax extension complete)
>>  same => n(hangup),Hangup()
>>
>> and the logs show that calling a fax using the voiceout extension in 
>> context test does not result in the fax extension being triggered.
>>
>> [Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Extension voiceout starting
>> [Jan 11 08:55:10] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Starting Answer Machine Detection
>> [Jan 11 08:55:13] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Answer Machine Detection 
>> MACHINE/LONGGREETING-1500-1500
>> [Jan 11 08:55:44] NOTICE[18073][C-0115] Ext. voicetest: 
>> SIP/31.13.156.183:5060-00f4: Voice out extension complete
>>
>> Just for completeness this is how the call is originated, with a 
>> different phone number:
>>
>> Action: Originate
>> ActionId: 1234567W001-125
>> Context: test
>> Exten: voicetest
>> Priority: 1
>> Channel: SIP/+441632660987@31.13.156.183:5060
>> Timeout: 6
>> Async: True
>>
>> Can anyone offer any insight into why this isn't working?
>>
>> Neil Youngman
>>
>>
> 
> You didn't ANSWER() the call

It's an outgoing call. I wouldn't expect to answer an outgoing call?

Neil

Neil Youngman 
Developer
Wirefast Limited

Wirefast provides secure corporate messaging services.
See our messaging solutions at  http://www.wirefast.com/ 
Please consider the environment.
Does this email or attachment need to be printed?
This message contains confidential information and is intended only 
for the individual named. If you are not the named addressee you 
should not disseminate, distribute or copy this email. Please 
notify the sender immediately by email if you have received this 
email by mistake and delete this email from your system.

Any views or opinions are solely those of the author
and do not necessarily represent those of Wirefast Limited

Email transmission cannot be guaranteed to be secure or error-free 
as information could be intercepted, corrupted, lost, destroyed, 
arrive late or incomplete, or contain viruses. The sender therefore 
does not accept liability for any errors or omissions in the contents 
of this message which arise as a result of email transmission.
Wirefast Limited is registered in England & Wales
Company number: 03865860
Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ

Wirefast definitions of classification can be found here:

Re: [asterisk-users] Freepbx / Asterisk PJsip multipe devices

2019-02-11 Thread Floimair Florian

Or just do it right using the PJSIP_DIALPLAN_CONTACTS function:

Dial(${PJSIP_DIAL_CONTACTS(${EXTEN})})

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 06.02.19, 15:26 schrieb "asterisk-users im Auftrag von Mitch Claborn" 
:

You can do this in the dial plan. Register the devices separately and 
include both addresses in the Dial() command.


Mitch

On 2/6/19 8:16 AM, basti wrote:
> In other words.
> 
> I there a way that both phones are ring with only one extension?
> 
> On 06.02.19 15:05, basti wrote:
>> both phones are in the same net.
>> when the soft phone is shut down, on hardware phone only an led is
>> flashing to show an incoming call but no sound.
>>
>> both phones use the same extension. that is the reason why I use pjsip.
>>
>> On 06.02.19 14:59, Antony Stone wrote:
>>> On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote:
>>>
 These two phones are not using the same extension, are they?
>>>
>>> If you shut down the softphone, does the hardware phone then ring?
>>>
>>>
>>> Antony.
>>>
 On 2/6/2019 8:49 AM, basti wrote:
> both phones are registered. and the hardware phone can also make 
calls.
> but an incoming call is not displayed and also not hearing.
>
> Call Waiting is also disabled.
>
> On 06.02.19 14:07, Cyril Alberts wrote:
>> Hi,
>> look at your registrations, is the hardware phone registered?
>> if yes, which phone vendor do you want to connect? can you make
>> outgoing calls with hardwarephone?
>>
>> BR Cyril
>>
>> Am Mittwoch, den 06.02.2019, 13:00 +0100 schrieb basti:
>>> Hello,
>>>
>>> I have some user that had have a hardwarephone and an softphone. I
>>> use pjsip driver and set "Max Contacts = 2" to have register both 
at the
>>> same time.
>>>
>>> But Only the softphone is ring. the hardware phone is mute.
>>>
>>> How can i fix this?
>>>
>>
> 

-- 
_
-- 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] trying to upgrade asterisk and Debian -- not working (John Covici)

2019-01-24 Thread Floimair Florian

You need to run
make uninstall_all
while you still have 13.24.0-rc1 checked out.
Then checkout the previous version, rebuild it and make install.
13.15.0 doesn't know anything about modules added by 13.24.0.
You usually would get a warning when running make install that there are 
modules present that were not compiled with the current version.

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 24.01.19, 08:52 schrieb "asterisk-users im Auftrag von John Covici" 
:

I checked out 13.15.0, ./configure, make delete all modules, followed
by make install.

On Thu, 24 Jan 2019 01:17:32 -0500,
Stefan Viljoen wrote:
> 
> What procedure did you follow to revert back to the old version?
> 
> It sounds like your binary has been revereted, but the modules it needs 
to load are still the 13.24.0-rc1 modules...
> 
> ---
> Hi.  I am trying to upgrade my asterisk from 13.15 to the latest of 
asterisk 13 which seems to be 13.24.0-rc1.  At the same time I want to go from 
Debian 8 to DEbian 9 to get a more recent operating system and applications.
> 
> I ran in to the following problems when trying to do this.
> 
> When trying to use asterisk 13.24.0-rc1, I ran into some strange problems 
with some of my custom scripts.
> 
> It seems the following statement immediately disconnects the user exten 
=> s,n,Read(digit,,1,,1,20) ; read a digit
> 
> In the log after that statement it says user disconnected.  I have an agi 
which speaks some text before the read and that agi does not even say anything, 
although it does complete.
> 
> Now, if I try to go back to 13.15.0, it does not work at all because it 
keeps telling in my log that modules support is not available, so no modules 
get loaded.
> 
> Any ideas on thispuzzle would be appreciated.
> 
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>  John Covici wb2una
>  cov...@ccs.covici.com
> 
> 
> 
> --
> 
> Subject: Digest Footer
> 
> ___
> --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
> 
> --
> 
> End of asterisk-users Digest, Vol 173, Issue 21
> ***
> 
> 
> -- 
> _
> -- 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
> 

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici wb2una
 cov...@ccs.covici.com

-- 
_
-- 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] Early Media Issue

2019-06-17 Thread Floimair Florian

Just a guess, but I suspect that this might be related with strictrtp setting 
in rtp.conf, which learns the correct source in doing so drops a few packets.
I would try to disable strictrtp for testing purposes and if this works add 
some delay before playing back the media.



With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com

Security and Communication by Commend

FN 178618z | LG Salzburg

Von: asterisk-users  im Auftrag von 
Mark Farmer 
Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum: Freitag, 14. Juni 2019 um 15:15
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [asterisk-users] Early Media Issue

Hi all

I've got an issue where when I call a number that just plays early media back 
to me.
Instead of hearing the full sequence of tones I hear a short ringing then part 
of the sequence. What seems odd is that I can see the telephone-event/8000 
being passed up the chain but when it gets to Asterisk, it is never sent back 
to the phone. Instead I just see the usual RTP flows.

I've been trying to fix this for hours, does anyone have any ideas how to get 
this working correctly?

Asterisk version is 13.25.0

The settings I think are relevant (I'm using chan_sip):

(sip.conf)
ignoresdpversion=yes
internal_timing=yes
progressinband=never
silencesuppression=no
prematuremedia=no

(Per peer)
progressinband=yes
directrtpsetup=no
dtmfmode=rfc2833
directmedia=no
silencesuppression=no
prematuremedia=no


TIA
Mark.

--
Mark Farmer
farm...@gmail.com
-- 
_
-- 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] h265 codec pass through on asterisk

2019-08-22 Thread Floimair Florian

Well, that sounds pretty straight forward.

I can do this and push it to gerrit.

Do I need to create a ticket for this?

 
 
With best regards

Florian Floimair
Innovation - Software-Development

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
http://www.commend.com 

Security and Communication by Commend

FN 178618z | LG Salzburg

Am 22.08.19, 11:55 schrieb "asterisk-users im Auftrag von Joshua C. Colp" 
:

On Thu, Aug 22, 2019, at 1:00 AM, mrcasa bengaluru wrote:
> 
> All,
> 
> I'm using asterisk 16.4.0 with h264 and opus quite well using linphone 
> 4.1 client on android and baresip on linux. 
> 
> I'm exploring use of h265 for improved video quality/lower network 
> bandwidth. I do not see pass through support on asterisk for h265/hvec. 
> All my SIP clients and underlying hardware have hvec/h265 encoding and 
> decoding available. 
> 
> I would have liked vp9 however, vp9 encoding in hardware is not 
widespread.
> 
> Is there any reason for not enabling support for h265 pass through?

The only real reason is that noone has added support for it. Asterisk has 
to be made aware of it in a few places in order to allow it. I did it for 
VP9[1] so that could be used as a base.

[1] 
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgerrit.asterisk.org%2Fc%2Fasterisk%2F%2B%2F6079data=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424140187sdata=ypez4T%2BQyKSK3MC7RYvPuPHlPw5Xjur8cE4tEh85BA0%3Dreserved=0

-- 
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: 
https://eur01.safelinks.protection.outlook.com/?url=www.digium.comdata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=O9zlOUmKWnhWRzClXD8VW7j67msa6g8WS2bgQTPDdl8%3Dreserved=0
 & 
https://eur01.safelinks.protection.outlook.com/?url=www.asterisk.orgdata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=PJ13owJbDqec%2FKrBE1mViLM97DMV4NH7Csg%2B90za5cs%3Dreserved=0

-- 
_
-- Bandwidth and Colocation Provided by 
https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.api-digital.comdata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=XRXWU%2FOD%2FOExpW9DzhWFN14lfepvV7x35owz8ZUzniA%3Dreserved=0
 --

Check out the new Asterisk community forum at: 
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fcommunity.asterisk.org%2Fdata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=nVs698D90CXkCFw%2BMa1NInm4UfxvlFYNnS7iB7XS8nI%3Dreserved=0

New to Asterisk? Start here:
  
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwiki.asterisk.org%2Fwiki%2Fdisplay%2FAST%2FGetting%2BStarteddata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=YsLAUmQvRDHQsnSH5isZbZUQGH%2FAvKnh%2BirqdpC6YyU%3Dreserved=0

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   
https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.digium.com%2Fmailman%2Flistinfo%2Fasterisk-usersdata=02%7C01%7Cf.floimair%40commend.com%7C0a9d13050a7d4d01b30f08d726e6e45c%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637020645424150178sdata=C7WcJxyOqPbvFBoQT3Ckxhw3FR1kwrQ76GVAJtNWNZw%3Dreserved=0

-- 
_
-- 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] Digium's Opus Codec download links broken?

2019-11-14 Thread Floimair Florian

Alright, seems to work again.
Thanks!

FLORIAN FLOIMAIR
Software Development - IMS
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
Phone: +43 662 85 62 25
Mail: f.floim...@commend.com
commend.com
LG Salzburg / FN 178618z



Von: asterisk-users  im Auftrag von 
"Joshua C. Colp" 
Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum: Donnerstag, 14. November 2019 um 14:51
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: Re: [asterisk-users] Digium's Opus Codec download links broken?

On Thu, Nov 14, 2019 at 9:46 AM Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
I tried to download Digium’s Opus Codec via the following link, but the server 
is unavailable: 
http://downloads.digium.com/pub/telephony/codec_opus/<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fdownloads.digium.com%2Fpub%2Ftelephony%2Fcodec_opus%2F=02%7C01%7Cf.floimair%40commend.com%7C71b4c7039aab4d18068c08d76909ad25%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637093362595572818=8PgUZbqh0HIPNSEbyYNwi9%2FYNxsrwj1myTP8ILIkSGU%3D=0>

It took me a while to figure this out, because initially I tried downloading 
via selecting the Opus codec in make menuselect and realizing that it isn’t 
there after make install step.

Can someone from Digium/Sangoma please confirm?


The server is there, but some individuals are having DNS problems with 
incorrect results. I've raised a ticket with Sangoma IT for it.

--
Joshua C. Colp
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: 
www.sangoma.com<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.sangoma.com%2F=02%7C01%7Cf.floimair%40commend.com%7C71b4c7039aab4d18068c08d76909ad25%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637093362595582813=TynHlbMpvF9kqahzd3PLLe2thuz%2FDS1TtB%2Bs37raYH8%3D=0>
 & 
www.asterisk.org<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.asterisk.org%2F=02%7C01%7Cf.floimair%40commend.com%7C71b4c7039aab4d18068c08d76909ad25%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637093362595582813=Mn1voCYlvGLaNxAxOJNsMIaLgoeiCr5ODOv62YH4oIM%3D=0>
-- 
_
-- 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

[asterisk-users] Digium's Opus Codec download links broken?

2019-11-14 Thread Floimair Florian

I tried to download Digium’s Opus Codec via the following link, but the server 
is unavailable: http://downloads.digium.com/pub/telephony/codec_opus/

It took me a while to figure this out, because initially I tried downloading 
via selecting the Opus codec in make menuselect and realizing that it isn’t 
there after make install step.

Can someone from Digium/Sangoma please confirm?




FLORIAN FLOIMAIR
Software Development - IMS
Commend International GmbH
Saalachstrasse 51
5020 Salzburg, Austria
Phone: +43 662 85 62 25
Mail: f.floim...@commend.com
commend.com
LG Salzburg / FN 178618z


-- 
_
-- 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] Length of dial string

2020-05-04 Thread Floimair Florian

Hi Paddy!

This used to be 80 characters total (including all characters like channel 
type, '&' and '/'. Had the same issue in the past where I extended that in the 
code and recompiled.
From what I understand there is basically no longer a hard limit in Dial since 
the recent change in the latest versions other than a single device must not 
exceed this but you can concatenate any number of them within the Dial string.

Hi all

as per the new release notice for 13.33.0 received today - can anyone advise
me the max limit of the string to the Dial Command - see 
*   [ASTERISK-27946

https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fissues.asterisk.org%2Fjira%2Fbrowse%2FASTERISK-27946data=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259sdata=JdT9Yvi7ml%2FqzIYMO39ks68rdMKY2P2DFIAGKCCh6a8%3Dreserved=0>
 ] - 
dial (API): Storage of dialed target uses AST_MAX_EXTENSION
when it shouldn't

I have been fighting with this issue for months trying to find a solution I
need to call 20+ devices at the same time so dial strings are very long I
cant really use a queue(ringall) which was my original idea as the customer
needs different groups for virtually every call some of which are simple sip
devices and others have to be local devices (Internal and External CLIs). 

Paddy Grice





-- 
_
-- Bandwidth and Colocation Provided by 
https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.api-digital.com%2Fdata=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259sdata=bGS2UUTgQGXjBEKqpzcXyJUAXXsap6PGIOHFTXCW4fc%3Dreserved=0
 --

Check out the new Asterisk community forum at: 
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fcommunity.asterisk.org%2Fdata=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259sdata=GrULRT%2Bgzqf6rCNUrcnYF81hvus3AWu%2BcLicHlOGFfE%3Dreserved=0

New to Asterisk? Start here:
  
https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwiki.asterisk.org%2Fwiki%2Fdisplay%2FAST%2FGetting%2BStarteddata=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259sdata=i41vU7Bo25YIliWGWPrrHrwDXodX83wcbG%2FIl%2FS5JIs%3Dreserved=0

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   
https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.digium.com%2Fmailman%2Flistinfo%2Fasterisk-usersdata=02%7C01%7Cf.floimair%40commend.com%7C5a5c413f7d8747dab6c408d7eda07497%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637239145718403259sdata=poTBflJdXsachl7MJCm5%2FUsL0V74bIHhou1Rhzedg2A%3Dreserved=0

-- 
_
-- 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] [External] 180 Ringing missing

2020-12-01 Thread Floimair Florian

If you have 183 Session progress, there is no need to send 180 Ringing 
(especially not AFTER 183 Session progress), as you already have early media 
instead. Having both is actually a bit misleading IMHO.

So this is actually correct. One should not rely on any of these 1xx 
"Provisional" messages.
They may or may not be sent, without violating SIP standards.

Am 01.12.20, 12:20 schrieb "asterisk-users im Auftrag von marek" 
:


hi,

after upgrade from Asterisk 11 to Asterisk 13.38.0(chan_sip) (i know,
its old. customer is very conservative...)

i have problem with missing 180 Ringing

flow is easy (PBX -> Asterisk -> SIP SBC)

Asterisk 11
PBX - Asterisk
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )
<- 180 Ringing
<- 200 OK

Asterisk 13
-> INVITE
<- 100 Trying
<- 183 Session Progress
( <- RTP -> )

__MISSING RINGING___

<- 200 OK

temporarily i solved problem with using "R" param

R: Default: Indicate ringing to the calling party, even if the called party
 isn't actually ringing. Allow interruption of the ringback if early
media
 is received on the channel.

it changed to

Asterisk 13 (Dial(${ARG1},300,R)
-> INVITE
<- 100 Trying
<- 180 Ringing
<- 183 Session Progress
( <- RTP -> )
<- 200 OK

any ideas why Ringing is missing? any solutions?

Marek



-- 
_
-- 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

[asterisk-users] Multiple 183 Session Progress for a single call

2020-12-17 Thread Floimair Florian

Hi List!

I am running into an „issue” that I cannot really explain.
I have a call from Station A to Station B with both legs connected to Asterisk 
via Kamailio.
Asterisk used is latest 18.1.0 with chan_pjsip.

Station A -> Kamailio -> Asterisk -> Kamailio -> Station B

Now when Station B gets the INVITE it answers with 183 Session Progress to 
initiate Early Media.
However Asterisk not only passes this 183 Session Progress on to Station A, it 
creates a second 183 Session Progress with basically the same
Content (only some Header lines might have switched position but their content 
is identical) to Station A.

The first question that comes to mind is:
Why does Asterisk send the second 183 Session Progress in the first place (it 
did not get anything from Station A or Station B that would explain this)?


And in turn this leads to an issue if I replace station A with another Asterisk 
and put station A behind this Asterisk.

Station A -> Asterisk2 -> Kamailio -> Asterisk -> Kamailio -> Station B

In that case whatever was in the SDP of the first 183 Session Progress (e.g. 
audio=recvonly video=inactive) will cause the second Asterisk to put both media 
streams to sendrecv, and starts sending RTP packets from Station A for both 
audio and video which clearly is a bug in Asterisk. By the way we tested with 
different versions from the latest 18 back to an early 16 version. They all 
show this behaviour.
To recap both 183 Session Progress contain the same SDP, the first one leads to 
wanted behaviour, the second one causes Asterisk to send both audio and video 
even though it shouldn’t.
-- 
_
-- 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] [External] Re: Multiple 183 Session Progress for a single call

2020-12-17 Thread Floimair Florian

Thanks Joshua for the quick answer!

I may put some investigation effort into this after the holidays regarding the 
known-issue.
For now we “help” ourselves by dropping the second 183 Session progress in 
Kamailio to mitigate the issue that arises in the second scenario.

Best regards!

Von: asterisk-users  im Auftrag von 
"Joshua C. Colp" 
Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum: Donnerstag, 17. Dezember 2020 um 15:22
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [External] Re: [asterisk-users] Multiple 183 Session Progress for a 
single call


CAUTION: This email originated from outside of the organization. Do not click 
links or open attachments unless you recognize the sender and know the content 
is safe.
On Thu, Dec 17, 2020 at 10:14 AM Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
Hi List!

I am running into an „issue” that I cannot really explain.
I have a call from Station A to Station B with both legs connected to Asterisk 
via Kamailio.
Asterisk used is latest 18.1.0 with chan_pjsip.

Station A -> Kamailio -> Asterisk -> Kamailio -> Station B

Now when Station B gets the INVITE it answers with 183 Session Progress to 
initiate Early Media.
However Asterisk not only passes this 183 Session Progress on to Station A, it 
creates a second 183 Session Progress with basically the same
Content (only some Header lines might have switched position but their content 
is identical) to Station A.

The first question that comes to mind is:
Why does Asterisk send the second 183 Session Progress in the first place (it 
did not get anything from Station A or Station B that would explain this)?

There is already an open issue for this[1], but noone has worked on it as of 
yet.

And in turn this leads to an issue if I replace station A with another Asterisk 
and put station A behind this Asterisk.

Station A -> Asterisk2 -> Kamailio -> Asterisk -> Kamailio -> Station B

In that case whatever was in the SDP of the first 183 Session Progress (e.g. 
audio=recvonly video=inactive) will cause the second Asterisk to put both media 
streams to sendrecv, and starts sending RTP packets from Station A for both 
audio and video which clearly is a bug in Asterisk. By the way we tested with 
different versions from the latest 18 back to an early 16 version. They all 
show this behaviour.
To recap both 183 Session Progress contain the same SDP, the first one leads to 
wanted behaviour, the second one causes Asterisk to send both audio and video 
even though it shouldn’t.

I can't say for certainty, but this is likely because of the way codec and 
stream negotiation in Asterisk works. Each leg is independent (negotiated 
between Asterisk and an endpoint), and the result of an outgoing leg (be it 
from a 200 OK or 183 Session Progress) is not forwarded to the calling side. 
There is continued work being done to improve this so in the future it could 
become smarter, but it is not yet there.

[1] 
issues.asterisk.org/jira/browse/ASTERISK-28185<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fissues.asterisk.org%2Fjira%2Fbrowse%2FASTERISK-28185=04%7C01%7Cf.floimair%40commend.com%7C581d516089c74644460b08d8a29722b2%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637438117312983378%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=e9eGkOstRp1NKZfY%2B315NUitngFoT%2FZ8WoIOd0BDcDk%3D=0>

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at 
www.sangoma.com<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.sangoma.com%2F=04%7C01%7Cf.floimair%40commend.com%7C581d516089c74644460b08d8a29722b2%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637438117312993368%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=SMKjlwzSYhg1NWEdQh3xX%2B5lWaRrQecP4Zpu%2B93e9tI%3D=0>
 and 
www.asterisk.org<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.asterisk.org%2F=04%7C01%7Cf.floimair%40commend.com%7C581d516089c74644460b08d8a29722b2%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637438117313003365%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C1000=De%2F%2FLAVHAPL2%2F7tg5n4DBQ%2F4ra%2BVX7kkGVrVFw3na50%3D=0>
-- 
_
-- 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

[asterisk-users] SIP INFO messages with Content-Type: application/media_control+xml

2021-10-13 Thread Floimair Florian

Hi all!

We have a WebRTC user-agent (using sip.js) that is giving me headache.

When a different user-agent calls this user-agent, we frequently see Asterisk 
generating SIP INFO messages with

Content-Type: application/media_control+xml
Content-Length: 178



  
   

  



In the payload, that is sent from Asterisk to the caller UA (the WebRTC client 
is the callee).

While this does not do any harm usually, I have no clue yet what causes 
Asterisk to generate these INFO messages.
I do know what they are for (https://datatracker.ietf.org/doc/html/rfc5168), 
but not why Asterisk is generating those or what might be the trigger.

Anyone have any hint?

Thanks and best regards


FLORIAN FLOIMAIR
Symphony Cloud Services (1568)

-- 
_
-- 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] [External] Re: SIP INFO messages with Content-Type: application/media_control+xml

2021-10-13 Thread Floimair Florian

Thanks Josh!

I was already suspecting something like this.

FLORIAN FLOIMAIR



Von: asterisk-users  im Auftrag von 
"Joshua C. Colp" 
Antworten an: Asterisk Users Mailing List - Non-Commercial Discussion 

Datum: Mittwoch, 13. Oktober 2021 um 15:12
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [External] Re: [asterisk-users] SIP INFO messages with Content-Type: 
application/media_control+xml


CAUTION: This email originated from outside of the organization. Do not click 
links or open attachments unless you recognize the sender and know the content 
is safe.
On Wed, Oct 13, 2021 at 10:05 AM Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
Hi all!

We have a WebRTC user-agent (using sip.js) that is giving me headache.

When a different user-agent calls this user-agent, we frequently see Asterisk 
generating SIP INFO messages with

Content-Type: application/media_control+xml
Content-Length: 178



  
   

  



In the payload, that is sent from Asterisk to the caller UA (the WebRTC client 
is the callee).

While this does not do any harm usually, I have no clue yet what causes 
Asterisk to generate these INFO messages.
I do know what they are for 
(https://datatracker.ietf.org/doc/html/rfc5168<https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fdatatracker.ietf.org%2Fdoc%2Fhtml%2Frfc5168=04%7C01%7Cf.floimair%40commend.com%7C348e1e7064704e31218e08d98e4b21d5%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637697275616090301%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000=aaQE409aZSFLW0IKu3zsz187t5MNwDd7q%2FXkZd7mbck%3D=0>),
 but not why Asterisk is generating those or what might be the trigger.

Anyone have any hint?

They will occur as a result of a video client requesting a full frame, and 
there's a few places in Asterisk where it will generate one in certain 
scenarios. Generally it comes from another client though. It can be received 
either using an incoming INFO request, or via RTCP through a PLI or FIR. If 
using PJSIP then the INFO request is only used for the H.264 codec if the 
WebRTC option is not enabled.

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at 
www.sangoma.com<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.sangoma.com%2F=04%7C01%7Cf.floimair%40commend.com%7C348e1e7064704e31218e08d98e4b21d5%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637697275616100293%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000=BbHjQrV3t1OAspzrXLr1YcQ7k6eJCgUUSVBB40Q4Ai0%3D=0>
 and 
www.asterisk.org<https://eur01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.asterisk.org%2F=04%7C01%7Cf.floimair%40commend.com%7C348e1e7064704e31218e08d98e4b21d5%7C13b1ddb756454e7fbe663171548559da%7C0%7C0%7C637697275616100293%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000=6AM%2BZg%2FL9A04sXw7ntgTjZNzxy4iZ6UJlWRL9bTxyhk%3D=0>
-- 
_
-- 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] [External] Re: Asterisk 16.23.0 doesn't respond anymore

2021-12-14 Thread Floimair Florian

Still this is a PITA. Just use your name so we know who we are talking to from 
the headers without looking at the body.
There should only ever be one "Administrator" on a mailing list, which is the 
admin of the list itself.
If you are admin in your own domain, we simply shouldn't care about.

So please change your name or use another email for this list.

FLORIAN FLOIMAIR
Symphony Cloud Services (1568)
 

Am 13.12.21, 23:39 schrieb "asterisk-users im Auftrag von Administrator" 
:

CAUTION: This email originated from outside of the organization. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe.

Hi Mark

Le 13/12/2021 à 22:53, Mark Murawski a écrit :
> Hi,
>
> 1) You should change your name on your email client so it doesn't say
> "Administrator"
>

And for the name, emails are always signed with first name ;)


-- 
_
-- 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

39 matches

Mail list logo