[asterisk-users] PJSIP wizard reload not reloading ?

2019-07-25 Thread Jean-Denis Girard
Hi list,

I'm having a strange problem when using pjsip wizard and reloading
("pjsip reload" on CLI): some data (specifically endpoint/pickup_group)
is not modified.

For example, initially I have empty pickup group:

tiare*CLI> pjsip show endpoint xxx
...
 pickup_group   :

...

Then, I add endpoint/pickup_group = 0,3 to pjsip_wizard.conf, and
reload: pickup_group remains empty.

Then, if I change the line in pjsip_wizard.conf to
endpoint/pickup_group = 0, 3
  ^ note the space here!
then reload, and I get what was expected:
tiare*CLI> pjsip show endpoint xxx
...
pickup_group   : 0, 3
...

I have seen this problem on Asterisk-16 only (up to latest 16.5.0).

The modified configuration file is included from
/etc/asterisk/pjsip_wizard.conf:
#include astportal/pjsip_wizard.conf

pjsip reload has default definition in cli_aliases.conf:
pjsip reload=module reload res_pjsip.so
res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so
res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so
res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so

Did I miss something, or should I open an issue?


Thanks,
-- 
Jean-Denis Girard

SysNux   Systèmes   Linux   en   Polynésie  française
https://www.sysnux.pf/   Tél: +689 40.50.10.40 / GSM: +689 87.797.527



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[asterisk-users] Male French Talent

2019-07-25 Thread Dovid Bender
Hi,

Does anyone out there know of male french talent for Asterisk sound files
where the talent already recorded the bulk of the Asterisk sound files?

TIA.

Regards,

Dovid
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Re: [asterisk-users] Wanted: professional softphone

2019-07-25 Thread Israel Gottlieb
look at zoiper
oem.zoiper.com
you could create a url that creates a build with all credentials
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[asterisk-users] Calls and queue statistics

2019-07-25 Thread Administrator TOOTAI

Hello list,

I'm looking for a solution that can be applied to a stock asterisk 16 
(pjsip if it matter) running Debian 9 (php7.0).


Statistics should be available for normal calls and queues using a WEB 
interface. Open source better but not necessary,


Any feedback appreciate, no matter if it's a "go for it" or "go away".

Regards

--
Daniel

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Re: [asterisk-users] Wanted: professional softphone

2019-07-25 Thread Carlos Chavez

On 7/24/19 11:41 PM, Michael Maier wrote:


Hello!

Does anybody by chance know of a softphone which additionally has a management 
suite to fully configure it userID based for Windows clients? Any idea is 
appreciated!

    Zulu from Sangoma allows you to generate a QR code that configures 
everything automatically for each user.  Been using it lately and it 
works very well.  Only downside is that it is only for FreePBX/PBXact.




--
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Carlos Chávez
+52 (55)8116-9161


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[asterisk-users] Asterisk 16.5.0 Now Available

2019-07-25 Thread Asterisk Development Team
The Asterisk Development Team would like to announce the release of Asterisk 
16.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.5.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
  no body causes crash
  (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
  reINVITE
  (Reported by Francesco Castellano)

Bugs fixed in this release:
---
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
  systems caused by ASTERISK-28317
  (Reported by abelbeck)
 * ASTERISK-28458 - res_pjsip_sdp_rtp: Remove unused variable
  
  (Reported by Michael Maier)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
  logs
  (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
  not logged
  (Reported by Bernhard Schmidt)
 * ASTERISK-28419 - app_amd: Does not work with silence
  suppression
  (Reported by Nasir Iqbal)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
  fragmentation on handshake server hello certificate
 
  (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
  Asterisk attempts to generate hangup event
  (Reported by
  Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
 
  (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
  gatewaying
  (Reported by pasandev)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-28421 - Wrong type used for timestamp in
  res_rtp_asterisk
  (Reported by Morten Tryfoss)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
  after Progress()
  (Reported by Gregory Massel)

Improvements made in this release:
---
 * ASTERISK-28234 - pbx_dundi: Add IPv4/IPv6 dual bind support
  for DUNDi
  (Reported by Kirsty Tyerman)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0

Thank you for your continued support of Asterisk!
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[asterisk-users] Asterisk 13.28.0 Now Available

2019-07-25 Thread Asterisk Development Team
The Asterisk Development Team would like to announce the release of Asterisk 
13.28.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.28.0 resolves several issues reported by the
community and would have not been possible without your participation.

Thank you!

The following issues are resolved in this release:

Security bugs fixed in this release:
---
 * ASTERISK-28447 - res_pjsip_messaging: In-dialog MESSAGE with
  no body causes crash
  (Reported by Gil Richard)
 * ASTERISK-28465 - Broken SDP can cause a segfault in a T.38
  reINVITE
  (Reported by Francesco Castellano)

Bugs fixed in this release:
---
 * ASTERISK-28457 - [patch] Fix crash in chan_dahdi on 32-bit
  systems caused by ASTERISK-28317
  (Reported by abelbeck)
 * ASTERISK-26006 - Show offending IP for TLS setup failures in
  logs
  (Reported by Oleksandr Natalenko)
 * ASTERISK-28444 - chan_pjsip: Peer IP for SSL handshake errors
  not logged
  (Reported by Bernhard Schmidt)
 * ASTERISK-28460 - res_pjsip_sdp_rtp: Fix ICE candidate leak
  with specific usage
  (Reported by Joshua C. Colp)
 * ASTERISK-28018 - IP Fragmentation happening instead of DTLS
  fragmentation on handshake server hello certificate
 
  (Reported by vijay kumar)
 * ASTERISK-25371 - Crash in hangup at chan_pjsip.c:1749 when
  Asterisk attempts to generate hangup event
  (Reported by
  Abhay Gupta)
 * ASTERISK-28435 - cdr_pgsql: Unix socket doesn't work
 
  (Reported by Dmitry Svyatogorov)
 * ASTERISK-27981 - res_fax: Fax session leak with fax
  gatewaying
  (Reported by pasandev)
 * ASTERISK-28419 - app_amd: Does not work with silence
  suppression
  (Reported by Nasir Iqbal)
 * ASTERISK-28427 - new mwi.h include missing from some dahdi
  source files, causes build failure
  (Reported by Guido
  Falsi)
 * ASTERISK-27994 - PJSIP: Early media ringback not indicated
  after Progress()
  (Reported by Gregory Massel)

For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.28.0

Thank you for your continued support of Asterisk!
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