Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

2020-02-04 Thread Administrator 
Done, ASTERISK-28725 
 but against 
version 16,7.0 and 13.30.0 as the new ones aren't available in the list


Le 04/02/2020 à 19:30, Joshua C. Colp a écrit :
On Tue, Feb 4, 2020 at 2:28 PM Administrator > wrote:


Hello,

we just installed the latest 13 and 16 version of asterisk and face
problem on incoming calls: they are ended like in Asterisk 16

[2020-02-04 19:19:48] ERROR[3768][C-0001]: stasis_bridges.c:199
bridge_topics_init: Bridge id initialization required
[2020-02-04 19:19:48] WARNING[3768][C-0001]: bridge.c:809
bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f:
Could not
initialize topics

Asterisk 13

[2020-02-04 19:16:14] ERROR[17133][C-002b] stasis_bridges.c:
Bridge
id initialization required
[2020-02-04 19:16:14] WARNING[17133][C-002b] bridge.c: Bridge
0e8746e2-43e4-4c7c-9df8-32d7fe6e3727: Could not initialize topics

Switching back to previous versions and problem disappear


Please file an issue on the issue tracker[1] with complete console 
output and dialplan if possible. This isn't something that has been 
seen in testing and there hasn't been any other reports.


[1] https://issues.asterisk.org/jira

--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com  and 
www.asterisk.org 



--
Daniel

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Re: [asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

2020-02-04 Thread Joshua C. Colp 
On Tue, Feb 4, 2020 at 2:28 PM Administrator  wrote:

> Hello,
>
> we just installed the latest 13 and 16 version of asterisk and face
> problem on incoming calls: they are ended like in Asterisk 16
>
> [2020-02-04 19:19:48] ERROR[3768][C-0001]: stasis_bridges.c:199
> bridge_topics_init: Bridge id initialization required
> [2020-02-04 19:19:48] WARNING[3768][C-0001]: bridge.c:809
> bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not
> initialize topics
>
> Asterisk 13
>
> [2020-02-04 19:16:14] ERROR[17133][C-002b] stasis_bridges.c: Bridge
> id initialization required
> [2020-02-04 19:16:14] WARNING[17133][C-002b] bridge.c: Bridge
> 0e8746e2-43e4-4c7c-9df8-32d7fe6e3727: Could not initialize topics
>
> Switching back to previous versions and problem disappear
>

Please file an issue on the issue tracker[1] with complete console output
and dialplan if possible. This isn't something that has been seen in
testing and there hasn't been any other reports.

[1] https://issues.asterisk.org/jira

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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[asterisk-users] Asterisk 13.31.0 and 16.8.0 - Bridge problem on incoming calls

2020-02-04 Thread Administrator 

Hello,

we just installed the latest 13 and 16 version of asterisk and face 
problem on incoming calls: they are ended like in Asterisk 16


[2020-02-04 19:19:48] ERROR[3768][C-0001]: stasis_bridges.c:199 
bridge_topics_init: Bridge id initialization required
[2020-02-04 19:19:48] WARNING[3768][C-0001]: bridge.c:809 
bridge_base_init: Bridge da3bd3d1-cdea-4a05-8b3d-0ded8c561c5f: Could not 
initialize topics


Asterisk 13

[2020-02-04 19:16:14] ERROR[17133][C-002b] stasis_bridges.c: Bridge 
id initialization required
[2020-02-04 19:16:14] WARNING[17133][C-002b] bridge.c: Bridge 
0e8746e2-43e4-4c7c-9df8-32d7fe6e3727: Could not initialize topics


Switching back to previous versions and problem disappear

--
Daniel

--
_
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 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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Re: [asterisk-users] Looking for sample hangup_handler_pop and _wipe using vars

2020-02-04 Thread Richard Mudgett 
>From the indicated wiki page:

*Pop a hangup handler off a channel and optionally push a replacement*

same => 
n,Set(CHANNEL(hangup_handler_pop)=[[[context,]exten,]priority[(arg1[,...][,argN])]]);

*Pop all hangup handlers off a channel and optionally push a replacement*

same => 
n,Set(CHANNEL(hangup_handler_wipe)=[[[context,]exten,]priority[(arg1[,...][,argN])]]);


What the syntax above is indicating is that the value to assign is
*optional*.  Thus to only pop or wipe hangup handlers you *do not* provide
a new handler to push.

[hdlr1]
exten = s,1,NoOp()
same = n,Return()

[hdlr2]
exten = s,1,NoOp()
same = n,Return()

[hdlr3]
exten = s,1,NoOp()
same = n,Return()

[my_new_handler_context]
exten = my_new_handler_exten,3,NoOp()
same = n,Return()

[my_current_context]
...
; This wipes all set hangup handlers because we are not providing an
optional replacement handler to push onto the empty stack.
same = n,NoOp() ; <-- At this point we may or may not have hangup handlers
set on the channel
same = n,Set(CHANNEL(hangup_handler_wipe)=)
same = n,NoOp() ; <-- At this point we have no hangup handlers on the
channel

; This wipes all set hangup handlers and pushes a new one onto the empty
stack.
same = n,NoOp() ; <-- At this point we may or may not have hangup handlers
set on the channel
same =
n,Set(CHANNEL(hangup_handler_wipe)=my_new_handler_context,my_new_handler_exten,3)
same = n,NoOp() ; <-- At this point we have only the one hangup handler we
just pushed: my_new_handler_context,my_new_handler_exten,3

; Hangup handler test dialplan
exten = 101,1,NoOp()
same = n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr2,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1)
; What is the hangup handler stack when the channel is in the Echo
application?
same = n,Echo()
same = n,Hangup()

exten = 102,1,NoOp()
same = n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr2,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1)
same =
n,Set(CHANNEL(hangup_handler_pop)=my_new_handler_context,my_new_handler_exten,3)
; What is the hangup handler stack when the channel is in the Echo
application?
same = n,Echo()
same = n,Hangup()

exten = 103,1,NoOp()
same = n,Set(CHANNEL(hangup_handler_push)=hdlr1,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr2,s,1)
same = n,Set(CHANNEL(hangup_handler_push)=hdlr3,s,1)
same =
n,Set(CHANNEL(hangup_handler_pop)=my_new_handler_context,my_new_handler_exten,3)
; What is the hangup handler stack when the channel is in the Echo
application?
same = n,Echo()
same = n,Hangup()

You can enter the test dialplan portion above.  Then call the test
extensions to see what is on the channel's hangup handler stack while the
channel is in the Echo application by using the command line commands
mentioned on the wiki page.

Richard

On Mon, Feb 3, 2020 at 7:26 PM David P  wrote:

> Please point me to samples of popping and wiping hangup handlers. I don't
> need to use the values returned; I just need to clear any handlers before I
> push a new one.
>
> It's not clear at
> https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers+Specification how
> to provide vars on the right-hand side.
>
> 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
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[asterisk-users] Asterisk 17.2.0 Now Available

2020-02-04 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
17.2.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 17.2.0 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:

New Features made in this release:
---
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
  / "maxredirs" doesn't do anything
  (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
  ability to match on source port
  (Reported by Sean Bright)

Bugs fixed in this release:
---
 * ASTERISK-28679 - stasis application is destroyed after its
  creation
  (Reported by Francois Blackburn)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
  (Reported
  by Ross Beer)
 * ASTERISK-28714 - REGRESSION: Feature
  subscription_persistence_recreate (ASTERISK-27759) Causes
  Segfaults
  (Reported by Ross Beer)
 * ASTERISK-28677 - CDR billsec is always 0 for transferred
  calls
  (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
  dialtone times out after 0:16:40
  (Reported by Andrew
  Siplas)
 * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
  translation' output
  (Reported by Sean Bright)
 * ASTERISK-24484 - Update documentation for statsd module -
  usage requirements unclear
  (Reported by Dan Jenkins)
 * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
  not available RAM
  (Reported by Kevin Flyn)
 * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
  whitespace appears empty in the dialplan
  (Reported by
  Frank Matano)
 * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
  ODBC storage enabled and realtime voicemail_data is used
 
  (Reported by Stas Kobzar)
 * ASTERISK-27622 - empty voicemail.conf required for ARA
  (realtime) voicemail to leave message
  (Reported by Jim Van
  Meggelen)
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
  event
  (Reported by Niksa Baldun)
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
  failure when using according to usage help
  (Reported by
  Cedric BASSAGET)
 * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
  support for hostnames
  (Reported by Joshua C. Colp)
 * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
  be present instead of just one
  (Reported by
  AvayaXAsterisk)
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
  application to return error and hangup
  (Reported by Corey
  Farrell)
 * ASTERISK-28507 - Wiki docs missing for MessageWaiting
 
  (Reported by David M. Lee)
 * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
  does not preserve XML  version number
 
  (Reported by Bryan Nelson)
 * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
  with concurrent command pri show span X
  (Reported by Dirk
  Wendland)
 * ASTERISK-28633 - stasis bridge topic leak
  (Reported by
  Joeran Vinzens)
 * ASTERISK-28492 - pjsip reload not reloading wizard
  endpoint/pickup_group endpoint/call_group
  (Reported by
  Jean-Denis Girard)
 * ASTERISK-28562 - SIP WSS message not processed until next
  frame arrives
  (Reported by Robert Sutton)
 * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
  it's supposed to due to invalid syntax
  (Reported by
  Richard Kenner)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
  readsql
  (Reported by Boris P. Korzun)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
  emitting a COLP
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
  a Byte order mark is present
  (Reported by Robin Leffmann)
 * ASTERISK-28625 - Playback of local files impacted by large
  media cache
  (Reported by Kevin Reeves)
 * ASTERISK-28664 - "trustrpid" is misspelled in
  sip_to_pjsip.py
  (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
  fails to deactivate CDR.
  (Reported by Frederic LE FOLL)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
  build on 17.0.0
  (Reported by George Joseph)
 * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
  non-existent media stream if codecs create additional streams
  and offer does not have them
  (Reported by nappsoft)
 * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
  with config option
  (Reported by Kevin Harwell)
 * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function

[asterisk-users] Asterisk 16.8.0 Now Available

2020-02-04 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
16.8.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 16.8.0 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:

New Features made in this release:
---
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
  / "maxredirs" doesn't do anything
  (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
  ability to match on source port
  (Reported by Sean Bright)

Bugs fixed in this release:
---
 * ASTERISK-28679 - stasis application is destroyed after its
  creation
  (Reported by Francois Blackburn)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
  (Reported
  by Ross Beer)
 * ASTERISK-28714 - REGRESSION: Feature
  subscription_persistence_recreate (ASTERISK-27759) Causes
  Segfaults
  (Reported by Ross Beer)
 * ASTERISK-28677 - CDR billsec is always 0 for transferred
  calls
  (Reported by Maciej Michno)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
  dialtone times out after 0:16:40
  (Reported by Andrew
  Siplas)
 * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
  translation' output
  (Reported by Sean Bright)
 * ASTERISK-24484 - Update documentation for statsd module -
  usage requirements unclear
  (Reported by Dan Jenkins)
 * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
  not available RAM
  (Reported by Kevin Flyn)
 * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
  whitespace appears empty in the dialplan
  (Reported by
  Frank Matano)
 * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
  ODBC storage enabled and realtime voicemail_data is used
 
  (Reported by Stas Kobzar)
 * ASTERISK-27622 - empty voicemail.conf required for ARA
  (realtime) voicemail to leave message
  (Reported by Jim Van
  Meggelen)
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
  event
  (Reported by Niksa Baldun)
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
  failure when using according to usage help
  (Reported by
  Cedric BASSAGET)
 * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
  support for hostnames
  (Reported by Joshua C. Colp)
 * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
  be present instead of just one
  (Reported by
  AvayaXAsterisk)
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
  application to return error and hangup
  (Reported by Corey
  Farrell)
 * ASTERISK-28507 - Wiki docs missing for MessageWaiting
 
  (Reported by David M. Lee)
 * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
  does not preserve XML  version number
 
  (Reported by Bryan Nelson)
 * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
  with concurrent command pri show span X
  (Reported by Dirk
  Wendland)
 * ASTERISK-28633 - stasis bridge topic leak
  (Reported by
  Joeran Vinzens)
 * ASTERISK-28492 - pjsip reload not reloading wizard
  endpoint/pickup_group endpoint/call_group
  (Reported by
  Jean-Denis Girard)
 * ASTERISK-28562 - SIP WSS message not processed until next
  frame arrives
  (Reported by Robert Sutton)
 * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
  it's supposed to due to invalid syntax
  (Reported by
  Richard Kenner)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
  readsql
  (Reported by Boris P. Korzun)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
  emitting a COLP
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
  a Byte order mark is present
  (Reported by Robin Leffmann)
 * ASTERISK-28664 - "trustrpid" is misspelled in
  sip_to_pjsip.py
  (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
  build on 17.0.0
  (Reported by George Joseph)
 * ASTERISK-28659 - res_pjsip_sdp_rtp: Bundle includes
  non-existent media stream if codecs create additional streams
  and offer does not have them
  (Reported by nappsoft)
 * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
  with config option
  (Reported by Kevin Harwell)
 * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
  fails to deactivate CDR.
  (Reported by Frederic LE FOLL)
 * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
  documentation
  (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28609 - Memory Leak in

[asterisk-users] Asterisk 13.31.0 Now Available

2020-02-04 Thread Asterisk Development Team 
The Asterisk Development Team would like to announce the release of Asterisk 
13.31.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 13.31.0 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:

New Features made in this release:
---
 * ASTERISK-17491 - CURLOPT() needs a "followlocation" parameter
  / "maxredirs" doesn't do anything
  (Reported by candrews)
 * ASTERISK-28639 - res_pjsip_endpoint_identifier_ip: Add
  ability to match on source port
  (Reported by Sean Bright)

Bugs fixed in this release:
---
 * ASTERISK-28679 - stasis application is destroyed after its
  creation
  (Reported by Francois Blackburn)
 * ASTERISK-28423 - ARI causes STASIS Deadlock
  (Reported
  by Ross Beer)
 * ASTERISK-28714 - REGRESSION: Feature
  subscription_persistence_recreate (ASTERISK-27759) Causes
  Segfaults
  (Reported by Ross Beer)
 * ASTERISK-28677 - CDR billsec is always 0 for transferred
  calls
  (Reported by Maciej Michno)
 * ASTERISK-28706 - silk 24hHz doesn't show up in 'core show
  translation' output
  (Reported by Sean Bright)
 * ASTERISK-24484 - Update documentation for statsd module -
  usage requirements unclear
  (Reported by Dan Jenkins)
 * ASTERISK-28702 - chan_dahdi: holding a channel via flash to
  dialtone times out after 0:16:40
  (Reported by Andrew
  Siplas)
 * ASTERISK-28695 - core: minmemfree watermark uses free RAM,
  not available RAM
  (Reported by Kevin Flyn)
 * ASTERISK-28693 - chan_sip: SIP MESSAGE beginning with a
  whitespace appears empty in the dialplan
  (Reported by
  Frank Matano)
 * ASTERISK-23739 - [patch]Segfault forwarding voicemail with
  ODBC storage enabled and realtime voicemail_data is used
 
  (Reported by Stas Kobzar)
 * ASTERISK-27622 - empty voicemail.conf required for ARA
  (realtime) voicemail to leave message
  (Reported by Jim Van
  Meggelen)
 * ASTERISK-28349 - Pause reason not reported in QueueMember AMI
  event
  (Reported by Niksa Baldun)
 * ASTERISK-21794 - CLI command 'realtime update2' syntax
  failure when using according to usage help
  (Reported by
  Cedric BASSAGET)
 * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document
  support for hostnames
  (Reported by Joshua C. Colp)
 * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can
  be present instead of just one
  (Reported by
  AvayaXAsterisk)
 * ASTERISK-28682 - app_record: Lack of `beep` audio file causes
  application to return error and hangup
  (Reported by Corey
  Farrell)
 * ASTERISK-28507 - Wiki docs missing for MessageWaiting
 
  (Reported by David M. Lee)
 * ASTERISK-27759 - res_pjsip_pubsub: Subscription persistence
  does not preserve XML  version number
 
  (Reported by Bryan Nelson)
 * ASTERISK-28605 - chan_dahdi: Deadlock in Hangup Scenarios
  with concurrent command pri show span X
  (Reported by Dirk
  Wendland)
 * ASTERISK-28633 - stasis bridge topic leak
  (Reported by
  Joeran Vinzens)
 * ASTERISK-28492 - pjsip reload not reloading wizard
  endpoint/pickup_group endpoint/call_group
  (Reported by
  Jean-Denis Girard)
 * ASTERISK-27243 - contrib: valgrind.supp doesn't suppress what
  it's supposed to due to invalid syntax
  (Reported by
  Richard Kenner)
 * ASTERISK-28497 - func_odbc: truncating Unicode string on
  readsql
  (Reported by Boris P. Korzun)
 * ASTERISK-28647 - chan_sip: RTP frames not transmitted after
  emitting a COLP
  (Reported by Jean Aunis - Prescom)
 * ASTERISK-28667 - Asterisk ignores parsing of config files if
  a Byte order mark is present
  (Reported by Robin Leffmann)
 * ASTERISK-28664 - "trustrpid" is misspelled in
  sip_to_pjsip.py
  (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28663 - jansson: Support old versions
 
  (Reported by Joshua C. Colp)
 * ASTERISK-28636 - app_chanisavail+cdr: ChanIsAvail sometimes
  fails to deactivate CDR.
  (Reported by Frederic LE FOLL)
 * ASTERISK-28604 - app_meetme, chan_ooh323 and cdr_mysql don't
  build on 17.0.0
  (Reported by George Joseph)
 * ASTERISK-28660 - res_fax: wrap Asterisk initiated negotiation
  with config option
  (Reported by Kevin Harwell)
 * ASTERISK-28628 - Debian 10.2: Warning when app_voicemail is
  compiling
  (Reported by Stanislav Abramenkov)
 * ASTERISK-28626 - Missing arguments in PJSIP_CONTACT function
  documentation
  (Reported by Pascal Cadotte Michaud)
 * ASTERISK-28651 - chan_sip logs errors on tx to non-existent
  TCP connections
  (Reported by Jaco Kroon)
 * ASTERISK-28502 -