Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

[George Joseph]

On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens 
wrote:

> Hello
>
> using pjproject 2.5.5
> using asterisk-certified-13.8-cert1
>

IIRC there were API changes in pjproject 2.5 that aren't accounted for in
asterisk 13.8.  Try pjproject 2.4.5 first and let's see if that works


>
> Compiled pjproject 2.5.5 with :
> ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
> --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound
> --disable-opencore-amr
>
> Compiled Asterisk 13 with
> ./configure --libdir=/usr/lib64
>
> All pjproject modules are selectable in menuselect, so here no problem.
>
> Modules are present in /usr/lib64/asterisk/module (see below).
>
> But when I start asterisk, I get a lot of errors concerning res_pjsip (see
> below) on the asterisk CLI.
>
> Anyone have some input on this ?
>
>
> Thanks.
>
> Kind regards.
>
>
>
>
> [root@sip admin]# ls /usr/lib64/asterisk/modules | grep pjsip
> chan_pjsip.so
> func_pjsip_aor.so
> func_pjsip_contact.so
> func_pjsip_endpoint.so
> res_pjsip_acl.so
> res_pjsip_authenticator_digest.so
> res_pjsip_caller_id.so
> res_pjsip_config_wizard.so
> res_pjsip_dialog_info_body_generator.so
> res_pjsip_diversion.so
> res_pjsip_dlg_options.so
> res_pjsip_dtmf_info.so
> res_pjsip_endpoint_identifier_anonymous.so
> res_pjsip_endpoint_identifier_ip.so
> res_pjsip_endpoint_identifier_user.so
> res_pjsip_exten_state.so
> res_pjsip_header_funcs.so
> res_pjsip_logger.so
> res_pjsip_messaging.so
> res_pjsip_multihomed.so
> res_pjsip_mwi_body_generator.so
> res_pjsip_mwi.so
> res_pjsip_nat.so
> res_pjsip_notify.so
> res_pjsip_one_touch_record_info.so
> res_pjsip_outbound_authenticator_digest.so
> res_pjsip_outbound_publish.so
> res_pjsip_outbound_registration.so
> res_pjsip_path.so
> res_pjsip_pidf_body_generator.so
> res_pjsip_pidf_digium_body_supplement.so
> res_pjsip_pidf_eyebeam_body_supplement.so
> res_pjsip_publish_asterisk.so
> res_pjsip_pubsub.so
> res_pjsip_refer.so
> res_pjsip_registrar_expire.so
> res_pjsip_registrar.so
> res_pjsip_rfc3326.so
> res_pjsip_sdp_rtp.so
> res_pjsip_send_to_voicemail.so
> res_pjsip_session.so
> res_pjsip_sips_contact.so
> res_pjsip.so
> res_pjsip_t38.so
> res_pjsip_transport_management.so
> res_pjsip_transport_websocket.so
> res_pjsip_xpidf_body_generator.so
>
>
> Asterisk CLI :
>
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_registrar.so': 
> /usr/lib64/asterisk/modules/res_pjsip_registrar.so:
> undefined symbol: ast_sip_location_retrieve_aor_contacts_nolock
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_registrar.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_path.so': 
> /usr/lib64/asterisk/modules/res_pjsip_path.so:
> undefined symbol: ast_sip_location_retrieve_aor
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_path.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_authenticator_digest.so':
> /usr/lib64/asterisk/modules/res_pjsip_authenticator_digest.so: undefined
> symbol: ast_sip_retrieve_auths
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_authenticator_digest.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_dialog_info_body_generator.so':
> /usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so:
> undefined symbol: ast_sip_pubsub_unregister_body_generator
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_dialog_info_body_generator.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_sdp_rtp.so': 
> /usr/lib64/asterisk/modules/res_pjsip_sdp_rtp.so:
> undefined symbol: ast_sip_session_unregister_supplement
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_sdp_rtp.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_publish_asterisk.so':
> /usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so: undefined
> symbol: ast_sip_register_publish_handler
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_publish_asterisk.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_send_to_voicemail.so':
> /usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so: undefined
> symbol: ast_sip_session_unregister_supplement
> [Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module
> 'res_pjsip_send_to_voicemail.so' could not be loaded.
> [Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: Error
> loading module 'res_pjsip_diversion.so': 
> /usr/lib64/asterisk/mo

Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Carlos Chavez]

On 8/15/16 3:16 PM, Jonas Kellens wrote:


Hello

after I have upgraded from Asterisk 12 to 
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved in 
MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - 
Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - 
Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - 
Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - 
Wrong password



Is this a known problem ??


Second question I have : can I get the complete list of columns that 
can be used in realtime database for sip peers somewhere (update for 
Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, 
dtlscafile, dtlssetup possible ??



The first thing you need to test is if you are properly loading the 
realtime data.  The best way would be to enable "rtcachefriends=yes" and 
then "sip show peer XXX load".  If you are not getting anything then 
there is a problem with your realtime setup.  I used realtime sip until 
13.7 before switching to PJSIP so it should work.


I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.


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+52 (55)9116-91161


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[asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

[Jonas Kellens]

Hello

after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 
none of my realtime SIP peers (saved in MySQL DB) register anymore.



[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5076' - Wrong 
password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5072' - Wrong 
password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5062' - Wrong 
password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 
handle_request_register: Registration from 
'' failed for '78.119.140.190:5060' - Wrong 
password



Is this a known problem ??


Second question I have : can I get the complete list of columns that can 
be used in realtime database for sip peers somewhere (update for Ast 13) 
? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup possible ??





Thanks for the help.


Kind regards.

Jonas.

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[asterisk-users] PJSIP, DAHDI and Fanvil phones

[Carlos Chavez]

I am having a problem with Fanvil phones (X3) when they make a call 
through DAHDI.  Pure SIP calls flow normally but when a call goes 
through a DANDHI interface to the PSTN we only get one way audio.  This 
is Asterisk 13.10.0 (bundled pjsip) and Dahdi 2.11.1 with an Openvox 
A400 card (4 port FXO).  We also have Aastra phones and those do not 
have any problem making callsto the PSTN.  All phones are on the 
internal network and there is no NAT.  If I configure a SIP trunk to 
PSTN audio works both ways, only when going through dahdi do we lose audio.


I have never used Fanvil before today so I really do not know their 
best configuration settings for Asterisk.  Has anyone experienced this 
problem with Fanvil phones?  Any recommendations?  A SIP debug show 
proper invites and the correct IP for both phone and Asterisk, RTP flows 
both ways between Asterisk and the phone but only outgoing audio (from 
phone) is heard and there is no incoming (from pstn).



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[asterisk-users] SIP 603 response when call is not answered

[Hooman Fazaeli]

Hi

I have noticed that asterisk returns 'SIP 603' when the called party does
not answer.

My test setup is simple: two SIP phones (extensions: 100 and 111)
registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds.
When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to
111 (expected) and a '603 Decline' response to 100 (unexpected &
misleading).
It seems to me that a'480 Temporarily unavailable' response is more
suitable in this situation.

Is this a normal behavior of asterisk or a bug?

Thanks.
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[asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

[Jonas Kellens]

Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1

Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr 
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound 
--disable-opencore-amr


Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so here no problem.

Modules are present in /usr/lib64/asterisk/module (see below).

But when I start asterisk, I get a lot of errors concerning res_pjsip 
(see below) on the asterisk CLI.


Anyone have some input on this ?


Thanks.

Kind regards.




[root@sip admin]# ls /usr/lib64/asterisk/modules | grep pjsip
chan_pjsip.so
func_pjsip_aor.so
func_pjsip_contact.so
func_pjsip_endpoint.so
res_pjsip_acl.so
res_pjsip_authenticator_digest.so
res_pjsip_caller_id.so
res_pjsip_config_wizard.so
res_pjsip_dialog_info_body_generator.so
res_pjsip_diversion.so
res_pjsip_dlg_options.so
res_pjsip_dtmf_info.so
res_pjsip_endpoint_identifier_anonymous.so
res_pjsip_endpoint_identifier_ip.so
res_pjsip_endpoint_identifier_user.so
res_pjsip_exten_state.so
res_pjsip_header_funcs.so
res_pjsip_logger.so
res_pjsip_messaging.so
res_pjsip_multihomed.so
res_pjsip_mwi_body_generator.so
res_pjsip_mwi.so
res_pjsip_nat.so
res_pjsip_notify.so
res_pjsip_one_touch_record_info.so
res_pjsip_outbound_authenticator_digest.so
res_pjsip_outbound_publish.so
res_pjsip_outbound_registration.so
res_pjsip_path.so
res_pjsip_pidf_body_generator.so
res_pjsip_pidf_digium_body_supplement.so
res_pjsip_pidf_eyebeam_body_supplement.so
res_pjsip_publish_asterisk.so
res_pjsip_pubsub.so
res_pjsip_refer.so
res_pjsip_registrar_expire.so
res_pjsip_registrar.so
res_pjsip_rfc3326.so
res_pjsip_sdp_rtp.so
res_pjsip_send_to_voicemail.so
res_pjsip_session.so
res_pjsip_sips_contact.so
res_pjsip.so
res_pjsip_t38.so
res_pjsip_transport_management.so
res_pjsip_transport_websocket.so
res_pjsip_xpidf_body_generator.so


Asterisk CLI :

[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_registrar.so': 
/usr/lib64/asterisk/modules/res_pjsip_registrar.so: undefined symbol: 
ast_sip_location_retrieve_aor_contacts_nolock
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_registrar.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_path.so': 
/usr/lib64/asterisk/modules/res_pjsip_path.so: undefined symbol: 
ast_sip_location_retrieve_aor
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_path.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_authenticator_digest.so': 
/usr/lib64/asterisk/modules/res_pjsip_authenticator_digest.so: undefined 
symbol: ast_sip_retrieve_auths
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_authenticator_digest.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_dialog_info_body_generator.so': 
/usr/lib64/asterisk/modules/res_pjsip_dialog_info_body_generator.so: 
undefined symbol: ast_sip_pubsub_unregister_body_generator
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_dialog_info_body_generator.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_sdp_rtp.so': 
/usr/lib64/asterisk/modules/res_pjsip_sdp_rtp.so: undefined symbol: 
ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_sdp_rtp.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_publish_asterisk.so': 
/usr/lib64/asterisk/modules/res_pjsip_publish_asterisk.so: undefined 
symbol: ast_sip_register_publish_handler
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_publish_asterisk.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_send_to_voicemail.so': 
/usr/lib64/asterisk/modules/res_pjsip_send_to_voicemail.so: undefined 
symbol: ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_send_to_voicemail.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_diversion.so': 
/usr/lib64/asterisk/modules/res_pjsip_diversion.so: undefined symbol: 
ast_sip_session_unregister_supplement
[Aug 14 12:02:32] WARNING[20712]: loader.c:1086 load_resource: Module 
'res_pjsip_diversion.so' could not be loaded.
[Aug 14 12:02:32] WARNING[20712]: loader.c:599 load_dynamic_module: 
Error loading module 'res_pjsip_dlg_options.so': 
/usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: 
ast_sip_sessio

[asterisk-users] Certified Asterisk 13.8-cert2 Now Available

[Asterisk Development Team]

The Asterisk Development Team has announced the release of Certified Asterisk 
13.8-cert2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/certified-asterisk

The release of Certified Asterisk 13.8-cert2 resolves several issues reported 
by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26280 - DNS lookups can block channel media paths
  (Reported by Mark Michelson)
 * ASTERISK-26132 - PJSIP: provide transport type with received
  messages (Reported by Scott Griepentrog)
 * ASTERISK-26237 - Fax is detected on regular calls. (Reported by
  Richard Mudgett)
 * ASTERISK-23013 - [patch] Deadlock between 'sip show channels'
  command and attended transfer handling (Reported by Ben
  Smithurst)
 * ASTERISK-26214 - Allow arbitrary time for fax detection to end
  on a channel (Reported by Richard Mudgett)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.8-cert2

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] DAHDI on CentOS 7

[Carlos Chavez]

On 8/15/16 11:04 AM, Eric Wieling wrote:


"make config" should also install the init script.

On 08/15/2016 11:36 AM, Jerry Geis wrote:

>On my Fedora 24 system, the "dahdi-tools" package contains an old-style
>init script  /etc/rc.d/init.d/dahdi, and this seems to work just 
fine with

>systemd. I realize that CentOS != Fedora but if you have or can find an
>init script for an older CentOS, it might work fine on CentOS 7. I 
can send
>you the script file that I have, but of course I can't guarantee it 
will

>work on CentOS.

seems to work. I copied the dahdi.init to /etc/rc.d/init.d/dahdi, run 
chkconfig dahdi on and reboot

and modules are now loaded...

Thanks,

Jerry


That part is broken for CentOS 7.  A "make config" will not install 
the init script, you need to do it by hand.


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[asterisk-users] How to remove unused custom hints?

[Tomas Holy]

Hello list members,
after programing of dialplan I have some messy Custom:hints which I can see in 
'devstate list'. I didn't find any possibility how to remove this hints from 
Asterisk and I want remove them. 


Can you help me with that, please? I tried search about that something in 
documentation or on Google, but I didn't find anything. 


asterisk*CLI> devstate list  
 
- 
--- Custom Device States  
-
 --- 
--- Name: 'Custom:queuememberCALLERID(num)'  State: 'RINGING' 
--- 
--- Name: 'Custom:queuememberh'  State: 'NOT_INUSE' 
--- 
--- Name: 'Custom:queuemembers'  State: 'INUSE' 
---
 

Thank you


Have a nice day!




S pozdravem

Tomáš Holý
INTERCONNECT s.r.o.
Zákaznická linka: +420 61333
TEL: +420 61321
FAX: +420 246063179
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Re: [asterisk-users] DAHDI on CentOS 7

[Eric Wieling]

"make config" should also install the init script.

On 08/15/2016 11:36 AM, Jerry Geis wrote:

>On my Fedora 24 system, the "dahdi-tools" package contains an old-style
>init script  /etc/rc.d/init.d/dahdi, and this seems to work just fine with
>systemd. I realize that CentOS != Fedora but if you have or can find an
>init script for an older CentOS, it might work fine on CentOS 7. I can send
>you the script file that I have, but of course I can't guarantee it will
>work on CentOS.

seems to work. I copied the dahdi.init to /etc/rc.d/init.d/dahdi, run 
chkconfig dahdi on and reboot

and modules are now loaded...

Thanks,

Jerry


On Mon, Aug 15, 2016 at 8:36 AM, Jerry Geis > wrote:


What is needed to get DAHDI to start up correctly on CentOS 7 and
systemd...
I am using DAHDI-linux-complete 2.11.1

I saw mention in my search that it has been fixed after 2.11.1 but
cannot find
what the fix is.

Thanks,

Jerry






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Re: [asterisk-users] DAHDI on CentOS 7

[Jerry Geis]

>On my Fedora 24 system, the "dahdi-tools" package contains an old-style
>init script  /etc/rc.d/init.d/dahdi, and this seems to work just fine with
>systemd. I realize that CentOS != Fedora but if you have or can find an
>init script for an older CentOS, it might work fine on CentOS 7. I can send
>you the script file that I have, but of course I can't guarantee it will
>work on CentOS.


seems to work. I copied the dahdi.init to /etc/rc.d/init.d/dahdi, run
chkconfig dahdi on and reboot
and modules are now loaded...

Thanks,

Jerry


On Mon, Aug 15, 2016 at 8:36 AM, Jerry Geis  wrote:

> What is needed to get DAHDI to start up correctly on CentOS 7 and
> systemd...
> I am using DAHDI-linux-complete 2.11.1
>
> I saw mention in my search that it has been fixed after 2.11.1 but cannot
> find
> what the fix is.
>
> Thanks,
>
> Jerry
>
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Re: [asterisk-users] Asterisk 14.0.0-beta1 Now Available

[Marcelo Terres]

I'm trying to compile it with unbound but I'm getting the following error:

"The UNBOUND installation appears to be missing or broken."

Ubuntu 14.04.5 LTS \n \l

root@rtc:/usr/local/src/asterisk-14.0.0-beta1# dpkg -l | grep -i unboun
ii  libunbound-dev:amd64 1.4.22-1ubuntu4.14.04.2
amd64static library, header files, and docs for
libunbound
ii  libunbound2:amd641.4.22-1ubuntu4.14.04.2
amd64library implementing DNS resolution and
validation

Any ideas?

Regards,
Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres


On Wed, Jul 27, 2016 at 6:02 PM, Asterisk Development Team
 wrote:
> The Asterisk Development Team has announced the first beta of
> Asterisk 14.0.0. This beta is available for immediate
> download at http://downloads.asterisk.org/pub/telephony/asterisk
>
> The release of Asterisk 14.0.0-beta1 resolves several issues reported by the
> community and would have not been possible without your participation.
> Thank you!
>
> The following are the issues resolved in this beta:
>
> New Features made in this release:
> ---
>  * ASTERISK-25904 - PJSIP: add contact.updated event (Reported by
>   Alexei Gradinari)
>  * ASTERISK-26058 - [Patch] Add uptime and last reloaded to
>   FullyBooted AMI event (Reported by Niklas Larsson)
>  * ASTERISK-25925 - Allow Early Bridges on ARI Dials (Reported by
>   Mark Michelson)
>  * ASTERISK-26068 - Multicast RTP Options (Reported by Mark
>   Michelson)
>  * ASTERISK-26042 - ARI: Allow downloading of the media associated
>   with a stored recording (Reported by Matt Jordan)
>  * ASTERISK-25425 - logger: Add JSON structured logging (Reported
>   by Matt Jordan)
>  * ASTERISK-25900 - PJSIP Endpoint IP Access Controls (Reported by
>   Alexei Gradinari)
>  * ASTERISK-25972 - res_pjsip_exten_state: Use body generator to
>   publish extension state (Reported by Richard Mudgett)
>  * ASTERISK-25889 - ARI: Add separate "create" and "dial"
>   operations for channels (Reported by Mark Michelson)
>  * ASTERISK-25803 - [patch] chan_sip: Optionally supply
>   fromuser/fromdomain in SIP dial string (Reported by Walter
>   Doekes)
>  * ASTERISK-24919 - res_pjsip_config_wizard: Ability to write
>   contents to file (Reported by Ray Crumrine)
>  * ASTERISK-25670 - Add regcontext to PJSIP (Reported by Daniel
>   Journo)
>  * ASTERISK-25660 - Add sipp-sendfax.xml and spandspflow2pcap.py to
>   contrib/scripts. (Reported by Walter Doekes)
>  * ASTERISK-25591 - [patch] Complete List of Header Files
>   (#include): iwyu (Reported by Alexander Traud)
>  * ASTERISK-25551 - [patch]Ability to add channel to an existing
>   bridge by specifying an existing channel prefix (Reported by
>   Alec Davis)
>  * ASTERISK-25419 - Dialplan Application for Integration of StatsD
>   (Reported by Ashley Sanders)
>  * ASTERISK-25549 - Confbridge: Add participant timeout option
>   (Reported by Mark Michelson)
>  * ASTERISK-24922 - ARI: Add the ability to intercept hold and
>   raise an event (Reported by Matt Jordan)
>  * ASTERISK-25479 - Allow CDR's to be modified before being
>   dispatched to engines (Reported by Jonh Wendell)
>  * ASTERISK-25480 - [patch]Add field PauseReason on
>   QueueMemberStatus (Reported by Rodrigo Ramirez Norambuena)
>  * ASTERISK-25377 - res_pjsip: Change default "From user" from UUID
>   to something more palatable (Reported by Mark Michelson)
>  * ASTERISK-25252 - ARI: Add the ability to manipulate log channels
>   (Reported by Matt Jordan)
>  * ASTERISK-25259 - chan_pjsip: Add rtptimeout support (Reported by
>   Joshua Colp)
>  * ASTERISK-25238 - ARI: Support push configuration (Reported by
>   Matt Jordan)
>  * ASTERISK-25173 - ARI: Add the ability to load/reload/unload an
>   Asterisk module (Reported by Matt Jordan)
>  * ASTERISK-25006 - [patch] Add support set character for quoted
>   identifiers  (Reported by Rodrigo Ramirez Norambuena)
>  * ASTERISK-23186 - [patch] Add usegmtime option to cel_pgsql
>   (Reported by Rodrigo Ramirez Norambuena)
>  * ASTERISK-24931 - dns: Add support for SRV records. (Reported by
>   Joshua Colp)
>  * ASTERISK-24834 - DNS Overhaul: Implement the proposed core API -
>   sync/async functions, resolver registration (Reported by Matt
>   Jordan)
>  * ASTERISK-24836 - DNS Overhaul: Write a Resolver Implementation
>   (Reported by Matt Jordan)
>  * ASTERISK-22591 - [patch]Prevent Asterisk from writing received
>   SMS content in log (Reported by Jan Juergens)
>  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
>   (Reported by Dwayne Hubbard)
>  * ASTERISK-24703 - ARI: Add the ability to "transfer" (redirect) a
>   channel (Reported by Matt Jordan)
>  * ASTERISK-24363 - [patch] A

Re: [asterisk-users] DAHDI on CentOS 7

[Greg Woods]

On Mon, Aug 15, 2016 at 6:36 AM, Jerry Geis  wrote:

> What is needed to get DAHDI to start up correctly on CentOS 7 and
> systemd...
>

On my Fedora 24 system, the "dahdi-tools" package contains an old-style
init script  /etc/rc.d/init.d/dahdi, and this seems to work just fine with
systemd. I realize that CentOS != Fedora but if you have or can find an
init script for an older CentOS, it might work fine on CentOS 7. I can send
you the script file that I have, but of course I can't guarantee it will
work on CentOS.

--Greg
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[asterisk-users] DAHDI on CentOS 7

[Jerry Geis]

What is needed to get DAHDI to start up correctly on CentOS 7 and systemd...
I am using DAHDI-linux-complete 2.11.1

I saw mention in my search that it has been fixed after 2.11.1 but cannot
find
what the fix is.

Thanks,

Jerry
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