Re: [asterisk-users] SIP reload not changing codecs

[Annus Fictus]

Hello,

on Asterisk 13.13.1 working correctly

Regards


El 27/02/2017 a las 10:59, Steve Edwards escribió:

Asterisk 13.3.2

I change the allowed codec from ulaw to g729 in sip.conf and enter 
'sip reload' on the console, but calls continue to use ulaw until 
restart.


Before reload:

lc10*CLI> sip show settings
Global Signalling Settings:
---
  Codecs: (ulaw)

lc10*CLI> core show channel SIP/poly-e637-
  NativeFormats: (ulaw)
WriteFormat: slin
 ReadFormat: ulaw

After reload:

lc10*CLI> sip show settings
Global Signalling Settings:
---
  Codecs: (g729)

lc10*CLI> core show channel SIP/poly-e637-0001
  NativeFormats: (ulaw)
WriteFormat: slin
 ReadFormat: ulaw

After restart:

lc10*CLI> sip show settings
Global Signalling Settings:
---
  Codecs: (g729)

lc10*CLI> core show channel SIP/poly-e637-
  NativeFormats: (g729)
WriteFormat: slin
 ReadFormat: g729

Why do I need to restart to get calls to actually use the new codec?




--
_
-- 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] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Annus Fictus]

And Microsip using PJSIP SIP stack :)


El 16/02/2017 a las 08:15, Jonathan H escribió:

Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/

On 16 February 2017 at 13:04, Max Grobecker
 wrote:

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:

Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
security reasons I have enable TLS on our Asterisk server to support TLS 
authentication, It works well with hard phones. Has anybody in this forum use 
SIP Soft phones with TLS authentication enabled? Any suggestions?



Thanks,
Motty





--
_
-- 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] Does HEP require PJSIP or does it also works with SIP ?

[Annus Fictus]

Hello,

I'm not totally sure but HEP permit SIP signaling and RTCP data capture 
only on PJSIP channels. For chan_sip you have to use captagent.


Regards


El 03/01/2017 a las 10:04, Olivier escribió:

Hello,

On a newly built Asterisk 13.13.1 system, I can't make HEP work with 
chan_sip (though I could make it work with PJSIP on another instance).


Looking either at [1] or hep.conf, I can't see anything meaning HEP 
requires PJSIP.


Before diging deeper, may I simply ask if HEP requires PJSIP or not ?
What about a line mentioning the answer in above documents (to keep 
other from wondering the same thing) ?


Best regards


[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_hep





-- 
_
-- 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] sorcery.conf mappings

[Annus Fictus]

Look at:

https://javiervalencia.net/2015/12/06/asterisk-en-realtime/

(Spanish)

Regards


El 09/11/2016 a las 17:06, Joshua Colp escribió:

On Wed, Nov 9, 2016, at 05:59 PM, Carlos Chavez wrote:

  Is there some documentation for all the available sorcery.conf
mappings for realtime?  Asterisk already includes some tables in the
database that are not enabled by default on the sorcery.conf like
transports and outbound registrations.  There are no examples in the
file on how to enable them.  Where can I find some documentation to
enable those mappings?

There's nothing currently to document this fully, the closest thing
would be the PJSIP realtime page[1]. I have created an issue to do some
documentation though[2]. Are there any in particular you are interested
in?

[1] https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime
[2] https://issues.asterisk.org/jira/browse/ASTERISK-26572




--
_
-- 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] Mysql PJSIP realtime > 13.10?

[Annus Fictus]

Hello Carlos,

I'm testing CentOS 7 ODBC packages with PJSIP Realtime without problems.

Maybe you use a different configuration?

Regards

El 12/09/2016 a las 16:01, Carlos Chavez escribió:


On 9/12/16 3:39 PM, George Joseph wrote:




On Mon, Sep 12, 2016 at 2:31 PM, George Joseph > wrote:




On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez
mailto:cur...@telecomabmex.com>> wrote:

Has anyone successfully used Mysql realtime PJSIP with
Asterisk 13.11?  I have tried 13.11, 13.11.1 and 13.11.2 but
I always get the following error now:

Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'qualify_timeout' cannot be type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'expiration_time' cannot be type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246
require_mysql: Possibly unsupported column type
'enum('yes','no')' on column 'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162
require_mysql: Realtime table general@ps_contacts: column
'via_port' cannot be type 'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411
register_aor_core: Unable to bind contact
'sip:2001@192.168.2.165:5060
;transport=udp' to AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165
:5060;transport=udp has been
deleted

Up until 13.10 everything was working despite the
warnings about field types.  Now my phones will not
register.  I can make calls but not receive.  All database
modifications are done through alembic so they are supposed
to be up to date.  The only way I can find to solve this
issue right now is to restore a 13.10 backup for both the
database and Asterisk.


res_config_mysql has been in "extended" support for some time now
and it's possible it just will no longer work.  We only test
alembic changes with postgres or odbc now.  Your best bet is to
convert to res_odbc.


Oh yeah, if you really do need res_config_mysql, go ahead and open an 
issue at issues.asterisk.org  and we'll 
take a look.  Since we don't test with it though, we might not notice 
if it gets broken again in the future unless someone reports it.



I have solved the problem for the moment by changing the 
ps_contacts table with the "recommendations" res_config_mysql is 
giving.  I just modified all the fields to varchar and now my phones 
are registering.  Obviously this is not a solution as the database 
needs to be modified by alembic on future versions and it will keep 
breaking.


I tried to migrate to res_config_odbc about 6 months ago but my 
Asterisk kept crashing.  I was told that the crashes were due to the 
version of ODBC distributed by CentOS 7 and that I would have to 
compile my own to be able to solve the issue.  Has this been solved?  
Is the RPM ODBC package included with CentOS 7 still bugged?  I try to 
avoid using packages not included in the distribution as they make 
upgrades a pain later on.

--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161




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

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
  http://www.asterisk.org/community/astricon-user-conference

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] Mysql PJSIP realtime > 13.10?

[Annus Fictus]

Hello,

is there any reason you don't use ODBC with MySQL?

Regards


El 12/09/2016 a las 15:14, Carlos Chavez escribió:
Has anyone successfully used Mysql realtime PJSIP with Asterisk 
13.11?  I have tried 13.11, 13.11.1 and 13.11.2 but I always get the 
following error now:


Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 
'qualify_timeout' cannot be type 'int(10)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 
'expiration_time' cannot be type 'bigint(20)' (need char)
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1246 
require_mysql: Possibly unsupported column type 'enum('yes','no')' on 
column 'authenticate_qualify'
[Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 
require_mysql: Realtime table general@ps_contacts: column 'via_port' 
cannot be type 'int(11)' (need char)
[Sep 12 14:42:35] ERROR[24498]: res_pjsip_registrar.c:411 
register_aor_core: Unable to bind contact 
'sip:2001@192.168.2.165:5060;transport=udp' to AOR '2001'
  == Contact 2001/sip:2001@192.168.2.165:5060;transport=udp has been 
deleted


Up until 13.10 everything was working despite the warnings about 
field types.  Now my phones will not register.  I can make calls but 
not receive.  All database modifications are done through alembic so 
they are supposed to be up to date.  The only way I can find to solve 
this issue right now is to restore a 13.10 backup for both the 
database and Asterisk.





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

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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 13 and WebRTC

[Annus Fictus]

Hello,

I mean a working configuration (SIP o PJSIP) without patches or code 
corrections.


Thank you

Regards


El 09/09/2016 a las 03:47, marek cervenka escribió:

using in production

last asterisk 13 + pjsip bundled + pjsip patch for RTP/SAVPF (search 
pjsip conf) + sipml5 version from roginvs


https://github.com/DoubangoTelecom/sipml5/pull/238


Dne 08/09/2016 v 23:36 Annus Fictus napsal(a):

Hello list,

before to lost my time, I'd like know if someone have a WebRTC 
working configuration on Asterisk 13.11.0 SIP or PJSIP channel.


Thank you

Regards









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

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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 13 and WebRTC

[Annus Fictus]

Hello list,

before to lost my time, I'd like know if someone have a WebRTC working 
configuration on Asterisk 13.11.0 SIP or PJSIP channel.


Thank you

Regards



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

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
 http://www.asterisk.org/community/astricon-user-conference

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 does Ast use IP vs FQDN for SIP header fields

[Annus Fictus]

hello,

try to add fromdomain=yourdomain

in your trunk configuration.

regards


El 04/08/2016 a las 21:20, Telium Technical Support escribió:


We are working with an ISP that needs Asterisk to place a FQDN name in 
the SIP ‘FROM’ and ‘INVITE’ fields – where Asterisk is currently using 
an IP address.  A SIP trace shows the following from my Asterisk box:


INVITE sip:62351155@1.1.1.1 SIP/2.0

Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK49f1d30e

From: "MYNAM" ;tag=as3d9596b0

We tried adding hosts file entries mapping these IP’s to hostname’s 
but Asterisk didn’t use them.  Can someone suggest how to do this?






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] PJSIP - State of the art

[Annus Fictus]

Hello,

I'd like share with you my tests about PJSIP channel with the aim of 
improving the functioning of the channel:


 * Multi domain support not work correctly:
   https://issues.asterisk.org/jira/browse/ASTERISK-26026
 * Different context subscribe for each endpoint not possible:
   https://issues.asterisk.org/jira/browse/ASTERISK-25471
 * BLF don't work correctly on my tests with X-Lite, BRIA, JiTSI. Only
   work partially with microsip but because this softphone use the same
   SIP STACK (PJSIP). I test BLF with the latest Asterisk version and
   latest  FreePBX version. The problem is when a softphone is on the
   phone on the other softphone appear off-line. On Jitsi not is
   possible know if a Endpoints is or not online

The main idea of the new channel was working on a multi-domain 
environment, have more then one device registered with same credentials 
and have more stability.


Be Better still with Asterisk 1.11.X?

Regards



-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] PJSIP defaults for endpoints when using realtime

[Annus Fictus]

with templates.

Regards


El 13/07/2016 a las 23:49, Carlos Chavez escribió:
Until Asterisk 11 I could use sip.conf to set defaults for all 
phones (language, dtmf, vmexten, etc) and just leave many fields in 
the database as NULL.  What would be the proper way to do this for 
Asterisk 13 and PJSIP?






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

[asterisk-users] Asterisk hep.conf

[Annus Fictus]

hello,

I'm trying to use Asterisk 13.9.1 with Homer SIP Capture Server.

My hep.conf Asterisk configuration is:

[general]
enabled = yes
capture_address=107.170.151.154:9060
;capture_password = foo
capture_id = 2464

SIP Signaling work correctly but no RTCP STATS arrive to Homer Server. 
On the Asterisk Console, many messages like this:


NOTICE[3739] res_hep.c: Unable to send packet: Address Family mismatch 
between source/destination


Any hint?

Regards

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

Re: [asterisk-users] Agents.conf Device_state

[Annus Fictus]

Hello,

I'm using 13.9.1 version

thank you for your answer.

Now working fine.

Regards


El 17/06/2016 a las 20:06, Richard Mudgett escribió:



On Fri, Jun 17, 2016 at 11:50 AM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello,

I think Device State for Agents don't work correctly

My configuration:

agents.conf

[general]

[agent](!)
autologoff=15
ackcall=no
acceptdtmf=#
wrapuptime=5000
musiconhold=default
recordagentcalls=no
custom_beep=beep

[2000](agent)
fullname=Fulano

[2001](agent)
fullname=Zutano

[2002](agent)
fullname=Mengano

queue.conf (Agents Related)

member => Agent/2000
member => Agent/2001
member => Agent/2002
member => PJSIP/1000

You didn't state which Asterisk version you are using.  However, since 
you reference a PJSIP
channel and are using the new agent.conf syntax, you must be using 
Asterisk 12+.  The
syntax for using an agent in queue.conf is has also changed since 
chan_agent no longer exists

in Asterisk 12+.

See 
https://www.asterisk-blog.com/2016/02/10/converting-from-chan_agent-to-app_agent_pool/


Richard




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Agents.conf Device_state

[Annus Fictus]

Hello,

I think Device State for Agents don't work correctly

My configuration:

agents.conf

[general]

[agent](!)
autologoff=15
ackcall=no
acceptdtmf=#
wrapuptime=5000
musiconhold=default
recordagentcalls=no
custom_beep=beep

[2000](agent)
fullname=Fulano

[2001](agent)
fullname=Zutano

[2002](agent)
fullname=Mengano

queue.conf (Agents Related)

member => Agent/2000
member => Agent/2001
member => Agent/2002
member => PJSIP/1000

Restarting Asterisk and then:

CLI> *queue show ventas*
ventas has 0 calls (max 50) in 'ringall' strategy (0s holdtime, 0s 
talktime), W:0, C:0, A:0, SL:0.0% within 120s

   Members:
  Agent/2002 (ringinuse disabled) (Invalid) has taken no calls yet
  Agent/2000 (ringinuse disabled) (Invalid) has taken no calls yet
  Agent/2001 (ringinuse disabled) (Invalid) has taken no calls yet
  PJSIP/1000 (ringinuse disabled) (Unavailable) has taken no calls yet
   No Callers

Any hint?

Regards


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Asterisk 13 BLF/Presence

[Annus Fictus]

Hello I would like to know if anyone has been able to set up a 
workingBLF/Presence configuration with PJSIP channel. If yes, please 
share the configuration and Softphones/Phones used. Thank you Regards


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk

[Annus Fictus]

Hello,

in which moment Asterisk leave to qualify the realtime endpoint? When 
you restart Asterisk?


On my asterisk 13.9.1, qualify on realtime endpoints works correctly. My 
sorcery.conf:


[res_pjsip]
endpoint=realtime,ps_endpoints
endpoint=config,pjsip.conf,criteria=type=endpoint
auth=realtime,ps_auths
auth=config,pjsip.conf,criteria=type=auth
aor=realtime,ps_aors
aor=config,pjsip.conf,criteria=type=aor
domain_alias=realtime,ps_domain_aliases
domain_alias=config,pjsip.conf,criteria=type=domain_alias
contact=realtime,ps_contacts
contact=config,pjsip.conf,criteria=type=contact

[res_pjsip_endpoint_identifier_ip]
identify=realtime,ps_endpoint_id_ips
identify=config,pjsip.conf,criteria=type=identify

Regards


El 13/06/2016 a las 14:16, Francisco Valentin Vinagrero escribió:


Hi,

Yes, we’re implementing the dialplan in realtime too.

Here the contents of sorcery.conf:

[res_pjsip]

endpoint=realtime,ps_endpoints

aor=realtime,ps_aors

contact=realtime,ps_contacts

[res_pjsip_endpoint_identifier_ip]

identify=realtime,ps_endpoint_id_ips

Cheers, Francisco.

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Annus 
Fictus

*Sent:* 13 June 2016 14:11
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 

*Subject:* Re: [asterisk-users] PJSIP does not qualify contacts after 
starting Asterisk


Hello Francisco,

you have to use:

extensions => odbc,asterisk

only if you want use dialplan in Realtime

can you share your sorcery.conf file?

Regards

El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribió:

Hi all,

(sending this again from the correct address)

I’m running Asterisk 13.8.0 (I need to check if that happens with
13.9.1 too when I have the time to build it) with PJSIP realtime
config.

I’ve defined several aors in the table ps_aors, like this (real
url replaced by myurl):

*CLI> pjsip show aor pbx-node-1

  Aor: 


Contact: 
  

 
=


  Aor: pbx-node-1 0

Contact:  pbx-node-1/sip:myurl:5060 771bf6a7d4 Created  
0.000


 ParameterName: ParameterValue

 ===

 authenticate_qualify : false

 contact  : sip:myurl:5060

default_expiration : 3600

 mailboxes :

 max_contacts : 0

maximum_expiration   : 7200

 minimum_expiration   : 60

 outbound_proxy   : sip:myurl:5060

 qualify_frequency: 30

 qualify_timeout  : 3.00

 remove_existing  : false

 support_path : false

So I think that those aors should be qualified automatically when
I run Asterisk, but if I do “/pjsip show contacts”/, I get that it
was just Created but not qualified:

*CLI> pjsip show contacts

  Contact: 
  


=

  Contact:  pbx-node-1/sip:myurl:5060 771bf6a7d4 Created   0.000

And not a single OPTIONS message if I take a trace…

If I want Asterisk to start sending OPTIONS, I need to do pjsip
reload and after that, they are qualified and their status changes
dynamically:

*CLI> pjsip show contacts

  Contact: 
  


=

  Contact:  pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail
8.833


The extconfig.conf file looks like this:

[settings]

ps_endpoints => odbc,asterisk

ps_auths => odbc,asterisk

ps_aors => odbc,asterisk

ps_domain_aliases => odbc,asterisk

ps_endpoint_id_ips => odbc,asterisk

ps_contacts => odbc,asterisk

extensions => odbc,asterisk

Any idea why I need to reload PJSIP if I want the aors to be
qualified?

Cheers, Francisco.







-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] PJSIP does not qualify contacts after starting Asterisk

[Annus Fictus]

Hello Francisco,

you have to use:

extensions => odbc,asterisk

only if you want use dialplan in Realtime

can you share your sorcery.conf file?

Regards

El 13/06/2016 a las 10:21, Francisco Valentin Vinagrero escribió:


Hi all,

(sending this again from the correct address)

I’m running Asterisk 13.8.0 (I need to check if that happens with 
13.9.1 too when I have the time to build it) with PJSIP realtime config.


I’ve defined several aors in the table ps_aors, like this (real url 
replaced by myurl):


*CLI> pjsip show aor pbx-node-1

  Aor:  



Contact:   
 


 = 



  Aor: pbx-node-1 0

Contact: pbx-node-1/sip:myurl:5060  771bf6a7d4 Created 0.000

 ParameterName: ParameterValue

 ===

 authenticate_qualify : false

 contact  : sip:myurl:5060

default_expiration : 3600

 mailboxes :

 max_contacts : 0

maximum_expiration   : 7200

 minimum_expiration   : 60

 outbound_proxy   : sip:myurl:5060

 qualify_frequency: 30

 qualify_timeout  : 3.00

 remove_existing  : false

 support_path : false

So I think that those aors should be qualified automatically when I 
run Asterisk, but if I do “/pjsip show contacts”/, I get that it was 
just Created but not qualified:


*CLI> pjsip show contacts

  Contact:   
 


=

  Contact: pbx-node-1/sip:myurl:5060771bf6a7d4 Created 0.000

And not a single OPTIONS message if I take a trace…

If I want Asterisk to start sending OPTIONS, I need to do pjsip reload 
and after that, they are qualified and their status changes dynamically:


*CLI> pjsip show contacts

  Contact:   
 


=

  Contact: pbx-node-1/sip:myurl.ch:5060771bf6a7d4 Avail 8.833

The extconfig.conf file looks like this:

[settings]

ps_endpoints => odbc,asterisk

ps_auths => odbc,asterisk

ps_aors => odbc,asterisk

ps_domain_aliases => odbc,asterisk

ps_endpoint_id_ips => odbc,asterisk

ps_contacts => odbc,asterisk

extensions => odbc,asterisk

Any idea why I need to reload PJSIP if I want the aors to be qualified?

Cheers, Francisco.





-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Fwd: PJSIP subscribe

[Annus Fictus]

I have seen the problem is with:

Jitsi, XLite, Bria 3.X

Grandstream Wave work correctly.

I have to test with hardphone.

Regards


El 09/06/2016 a las 01:11, George Joseph escribió:



On Wed, Jun 8, 2016 at 8:48 AM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello,

How can I know if is a BUG and report on Asterisk-Jira?

Asterisk is sending the correct "ep:busy" status but the clients 
appear to not be using it.  I'm not sure what else we can do on the 
Asterisk side.


Thank you

Regards


 Mensaje reenviado 
Asunto: PJSIP subscribe
Fecha:  Mon, 6 Jun 2016 19:13:35 +0200
De: Annus Fictus 
<mailto:annusfic...@gmail.com>
Para:   Asterisk Users Mailing List - Non-Commercial Discussion

<mailto:asterisk-users@lists.digium.com>



Hello,

I'm trying to use presence with PJSIP and  I have a "issue".

I created correctly hint priorities like:

exten => 1000,hint,PJSIP/1000
exten => 1001,hint,PJSIP/1001

Extension 1000 can subscribe extension 1001 y vice-versa. The problem is
when the extension 1000 make or receive a call. In the softphone where
the extension is present on buddy list, the extension appear go offline.
When hang-up the extension return on-line.

On the Asterisk console the command core show hints show the correct
state for the extension.

On my tests I used XLite, Bria and Jitsi.

Any hint?

Thank you

Regards.



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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




--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com <http://www.digium.com/> & 
www.asterisk.org <http://www.asterisk.org/>






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Fwd: PJSIP subscribe

[Annus Fictus]

Hello,

How can I know if is a BUG and report on Asterisk-Jira?

Thank you

Regards


 Mensaje reenviado 
Asunto: PJSIP subscribe
Fecha:  Mon, 6 Jun 2016 19:13:35 +0200
De: Annus Fictus 
Para: 	Asterisk Users Mailing List - Non-Commercial Discussion 





Hello,

I'm trying to use presence with PJSIP and  I have a "issue".

I created correctly hint priorities like:

exten => 1000,hint,PJSIP/1000
exten => 1001,hint,PJSIP/1001

Extension 1000 can subscribe extension 1001 y vice-versa. The problem is
when the extension 1000 make or receive a call. In the softphone where
the extension is present on buddy list, the extension appear go offline.
When hang-up the extension return on-line.

On the Asterisk console the command core show hints show the correct
state for the extension.

On my tests I used XLite, Bria and Jitsi.

Any hint?

Thank you

Regards.


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] PJSIP subscribe

[Annus Fictus]

Hello,

thank you for the answer... how can I see the correct status?

any configuration on asterisk or softphone side?

Regards

El 07/06/2016 a las 16:36, George Joseph escribió:
I can confirm that Bria shows offline but if the client is using the 
tuple status instead of the person status then I can see why.



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

[asterisk-users] PJSIP subscribe

[Annus Fictus]

Hello,

I'm trying to use presence with PJSIP and  I have a "issue".

I created correctly hint priorities like:

exten => 1000,hint,PJSIP/1000
exten => 1001,hint,PJSIP/1001

Extension 1000 can subscribe extension 1001 y vice-versa. The problem is 
when the extension 1000 make or receive a call. In the softphone where 
the extension is present on buddy list, the extension appear go offline. 
When hang-up the extension return on-line.


On the Asterisk console the command core show hints show the correct 
state for the extension.


On my tests I used XLite, Bria and Jitsi.

Any hint?

Thank you

Regards.



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

[asterisk-users] JABBER_RECEIVE timeout don't work

[Annus Fictus]

Hello,

I'm trying to use JABBER_RECEIVE function on my dialplan but the timeout 
function don't work.


This is my dialplan:

[google-in]
exten => s,1,NoOp( Call from Gtalk )
same => n,SendText(Hola,Como te llamas?)
same => n,Set(nombre=${JABBER_RECEIVE(google,${CALLERID(name)},30)})
same => n,SendText(Hola ${nombre}, bienvenido en XYZ)
same => n,Set(CALLERID(name)=${nombre})
same => n,Wait(2)
same => n,SendText(Espera un momento mientras te comunicamos)
same => n,Dial(PJSIP/1000,30)
same => n,Hangup()

In theory the function have to wait a answer for 30 seconds but don't.  
Asterisk execute next dialplan line immediately.




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

Re: [asterisk-users] Asterisk PJSIP Multi-tenant

[Annus Fictus]

Done.

ASTERISK-26026

El 16/05/2016 a las 14:40, George Joseph escribió:



On Sun, May 15, 2016 at 10:17 PM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello,

with qualify_frequency=0 I can't receive calls from others endpoints.

Other strange think is if I set mailboxes parameter on the
console, when the endpoint registering, i can see:

ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable
to create outbound NOTIFY request to endpoint 1...@sip.domain.com
<mailto:1...@sip.domain.com>
WARNING[2208]: res_pjsip_mwi.c:379
send_unsolicited_mwi_notify_to_contact: Unable to create
unsolicited NOTIFY request to endpoint 1...@sip.domain.com
<mailto:1...@sip.domain.com> URI
sip:1001@95.250.29.3:50673;rinstance=1af959e7c0059fc4

Unsolicited NOTIFY and OPTIONS both use the out-of-dialog path so I'm 
guessing we have an issue there.  Open an issue at 
https://issues.asterisk.org




Regards

El 16/05/2016 a las 02:52, George Joseph escribió:



On Sun, May 15, 2016 at 12:00 PM, Annus Fictus
mailto:annusfic...@gmail.com>> wrote:

Hello List,

following this thread:


http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains

I tried to configure on the pjsip.conf the same endpoint with
different domains like:

[1...@sip.domain.com <mailto:1...@sip.domain.com>]
type=endpoint

[1...@sip1.domain.com <mailto:1...@sip1.domain.com>]
type=endpoint

I can register the two 1000 endpoints using different domain
but on the Asterisk console:

ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request:
Unable to create outbound OPTIONS request to endpoint
1...@sip.domain.com <mailto:1...@sip.domain.com>

ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact:
Unable to create request to qualify contact
sip:1000@95.250.29.3:53570
<http://sip:1000@95.250.29.3:53570>;rinstance=d90827763e4353c0

in the aor section I'm using:

qualify_frequency=30


If you set qualify_frequency=0, can that endpoint make and
receive calls?  Not suggesting this as a solution, just asking to
narrow possibilities down.


Any hint?

Regards


-- 
_

-- Bandwidth and Colocation Provided by
http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
every Thurs:
http://www.asterisk.org/hello

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




-- 
George Joseph

Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com <http://www.digium.com> &
www.asterisk.org <http://www.asterisk.org>






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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




--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com <http://www.digium.com/> & 
www.asterisk.org <http://www.asterisk.org/>






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Asterisk PJSIP Multi-tenant

[Annus Fictus]

Hello,

with qualify_frequency=0 I can't receive calls from others endpoints.

Other strange think is if I set mailboxes parameter on the console, when 
the endpoint registering, i can see:


ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to 
create outbound NOTIFY request to endpoint 1...@sip.domain.com
WARNING[2208]: res_pjsip_mwi.c:379 
send_unsolicited_mwi_notify_to_contact: Unable to create unsolicited 
NOTIFY request to endpoint 1...@sip.domain.com URI 
sip:1001@95.250.29.3:50673;rinstance=1af959e7c0059fc4


Regards

El 16/05/2016 a las 02:52, George Joseph escribió:



On Sun, May 15, 2016 at 12:00 PM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello List,

following this thread:


http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains

I tried to configure on the pjsip.conf the same endpoint with
different domains like:

[1...@sip.domain.com <mailto:1...@sip.domain.com>]
type=endpoint

[1...@sip1.domain.com <mailto:1...@sip1.domain.com>]
type=endpoint

I can register the two 1000 endpoints using different domain but
on the Asterisk console:

ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable
to create outbound OPTIONS request to endpoint 1...@sip.domain.com
<mailto:1...@sip.domain.com>

ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable
to create request to qualify contact sip:1000@95.250.29.3:53570
<http://sip:1000@95.250.29.3:53570>;rinstance=d90827763e4353c0

in the aor section I'm using:

qualify_frequency=30


If you set qualify_frequency=0, can that endpoint make and receive 
calls?  Not suggesting this as a solution, just asking to narrow 
possibilities down.



Any hint?

Regards


-- 
_

-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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




--
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com <http://www.digium.com/> & 
www.asterisk.org <http://www.asterisk.org/>






-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Asterisk PJSIP Multi-tenant

[Annus Fictus]

Hello List,

following this thread:

http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains

I tried to configure on the pjsip.conf the same endpoint with different 
domains like:


[1...@sip.domain.com]
type=endpoint

[1...@sip1.domain.com]
type=endpoint

I can register the two 1000 endpoints using different domain but on the 
Asterisk console:


ERROR[1748]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to 
create outbound OPTIONS request to endpoint 1...@sip.domain.com


ERROR[1748]: res_pjsip/pjsip_options.c:350 qualify_contact: Unable to 
create request to qualify contact 
sip:1000@95.250.29.3:53570;rinstance=d90827763e4353c0


in the aor section I'm using:

qualify_frequency=30

Any hint?

Regards


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

Re: [asterisk-users] Statsd Dialplan Application

[Annus Fictus]

Thank you for the response.

Regards

El 19/01/2016 a las 11:22, Kevin Harwell escribió:



On Tue, Jan 19, 2016 at 8:46 AM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello,

I'd like to do some tests with the StatsD dialplan application but
on the last version of Asterisk 13 (13.7.0) I can't find this
application.

New Features made in this release:
---
 * ASTERISK-25419 - Dialplan Application for Integration of StatsD
  (Reported by Ashley Sanders)

res_statsd module are correctly compiled y loaded.

Any hint?

Regards




Unfortunately these changes did not go out with the latest release of 
13.7.0. Actually the new StatsD Dialplan application currently resides 
in master only. A small change to the res_statsd api was made and got 
tagged with that issue number for some reason, thus making it look as 
if the StatsD application feature was added to 13.



--
Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:http://digium.com  &http://asterisk.org




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] Statsd Dialplan Application

[Annus Fictus]

Hello,

I'd like to do some tests with the StatsD dialplan application but on 
the last version of Asterisk 13 (13.7.0) I can't find this application.


New Features made in this release:
---
 * ASTERISK-25419 - Dialplan Application for Integration of StatsD
  (Reported by Ashley Sanders)

res_statsd module are correctly compiled y loaded.

Any hint?

Regards



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

Re: [asterisk-users] Help with CDR-Stats

[Annus Fictus]

CDR-STATS is for reporting.

A2Billing is for billing...

Regards

El 16/12/2015 a las 11:15, Vitor Mazuco escribió:

Hi everyone!

I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult.

Is there others optins for billing?

Thanks




--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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

Re: [asterisk-users] Issues with Twilio number incoming call and context matching

[Annus Fictus]

Maybe is because now it's a different context:

from-twilio-remove-plus

before

from-internal

is right?

regards

El 02/12/2015 a las 10:22, Sonny Rajagopalan escribió:
Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, 
TWILIO)). It does not work and NO error message in CLI.


I have also tried 
http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ 
since I first emailed this group, but that does not seem to work either.


Here is my log:

[Dec  2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: 
Call from 'from-twilio' (UDP:mysillyApp.pstn.twilio.com:5060 
<http://mysillyApp.pstn.twilio.com:5060>) to extension '+17775551212' 
rejected because extension not found in context 'from-twilio-remove-plus'.
[Dec  2 15:09:29] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: 
Call from 'from-twilio' (UDP:mysillyApp.pstn.twilio.com:5060 
<http://mysillyApp.pstn.twilio.com:5060>) to extension '+17775551212' 
rejected because extension not found in context 'from-twilio-remove-plus'.

ip-*CLI> dialplan show from-twilio-remove-plus
[ Context 'from-twilio-remove-plus' created by 'pbx_config' ]
'_[+1]XX' => 1. GotoIf($["${CALLERID(num):0:2}" != 
"+1"]?noplusatstart) [pbx_config]

2. Log(WARNING,TWILIO)  [pbx_config]
3. Set(CALLERID(num)=${CALLERID(num):1})  [pbx_config]
 [noplusatstart] 4. Goto(from-external,${EXTEN},1)   [pbx_config]
ip-*CLI> dialplan show from-external
[ Context 'from-external' created by 'pbx_config' ]
'17775551212' =>  1. Log(WARNING,TWILIO)[pbx_config]
2. Hangup() [pbx_config]


On Wed, Dec 2, 2015 at 9:23 AM, Annus Fictus <mailto:annusfic...@gmail.com>> wrote:


Hello,

try to change:

exten => 17775551212,1,Log(WARNING, TWILIO)
same =>  n,Hangup()

with:

exten => +17775551212,1,Log(WARNING, TWILIO)
same =>n,Hangup()

Regards

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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 --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] Issues with Twilio number incoming call and context matching

[Annus Fictus]

Hello,

try to change:

exten => 17775551212,1,Log(WARNING, TWILIO)
same =>  n,Hangup()

with:

exten => +17775551212,1,Log(WARNING, TWILIO)
same =>  n,Hangup()

Regards
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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