[asterisk-users] how do I run a command on "Failed to authenticate" ?

2020-09-11 Thread sean darcy

16.13.0, pjsip

I'd like to get an alert if a call fails to authenticate:

if "Failed to authenticate" then
   mail someone the source ip
endif

As I look at ami or ari, they deal with calls in channels. Is there a 
way to get failed invites or registers ?


sean



--
_
-- 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] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

2020-09-10 Thread sean darcy

On 9/7/20 3:41 PM, Joshua C. Colp wrote:
On Sat, Sep 5, 2020 at 10:23 AM sean darcy <mailto:seandar...@gmail.com>> wrote:





module load res_pjsip
Unable to load module res_pjsip
Command 'module load res_pjsip' failed.
ERROR[141535]: loader.c:281 module_load_error: Error loading module
'res_pjsip': /usr/lib64/asterisk/modules/res_pjsip.so: undefined
symbol:
ast_statsd_log_full_va

module load chan_pjsip
Unable to load module chan_pjsip
Command 'module load chan_pjsip' failed.
ERROR[141780]: loader.c:281 module_load_error: Error loading module
'chan_pjsip': /usr/lib64/asterisk/modules/chan_pjsip.so: undefined
symbol: ast_sip_cli_traverse_objects

/usr/include/asterisk exists, with all the .h files, owned by root,
permissions 644. For instance:

grep ast_sip_cli_traverse_objects /usr/include/asterisk/*
...
/usr/include/asterisk/res_pjsip_cli.h:char
*ast_sip_cli_traverse_objects(struct ast_cli_entry *e, int cmd, struct
ast_cli_args *a);


Do I need to preload some module?

Any help appreciated.


Your PJSIP has built requiring the res_statsd module, loading that 
before res_pjsip should allow it to load. If not you'd need to provide 
the new output.


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


The problem was that I was not building app_statsd so the res_statsd 
wasn't built.


I can't figure out where I configured pjproject to use statsd.

sean


--
_
-- 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] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

2020-09-07 Thread sean darcy

On 9/6/20 3:43 AM, Michael Maier wrote:

On 05.09.20 at 15:22 sean darcy wrote:

asterisk-16.13.0-rc2. Fedora 32

pjsip won't load because of undefined symbols:


This means, that your pjsip library doesn't match the asterisk binary. It's 
best to remove the independent pjsip library and compile asterisk[1] with the 
bundled pjsip library. Doing
it this way ensures that pjsip and asterisk match for sure (and some additional 
patches are applied to pjsip on top regarding usage of pjsip in asterisk).


Greetings
Michael

[1] https://downloads.asterisk.org/pub/telephony/asterisk


Sure seems I'm using the bundled pjproject:

./bootstrap.sh
Generating the configure script for Asterisk ...
..
./configure --build=x86_64-redhat-linux-gnu 
--host=x86_64-redhat-linux-gnu --program-prefix= 
--disable-dependency-tracking --prefix=/usr --exec-prefix=/usr 
--bindir=/usr/bin --sbindir=/usr/sbin --sysconfdir=/etc 
--datadir=/usr/share --includedir=/usr/include --libdir=/usr/lib64 
--libexecdir=/usr/libexec --localstatedir=/var --sharedstatedir=/var/lib 
--mandir=/usr/share/man --infodir=/usr/share/info --with-imap=system 
--with-gsm=/usr --with-ilbc=/usr --with-libedit=yes --with-srtp 
--with-pjproject-bundled 'LDFLAGS=-m64 
-Wl,--as-needed,--library-path=/usr/lib64 -Wl,-z,relro -Wl,--as-needed 
-Wl,-z,now -specs=/usr/lib/rpm/redhat/redhat-hardened-ld'

...
checking for embedded pjproject (may have to download)... configuring
[pjproject]  Downloading 
https://raw.githubusercontent.com/asterisk/third-party/master/pjproject/2.10/pjproject-2.10.tar.bz2 
to /tmp/pjproject-2.10.tar.bz2

[pjproject]  Verifying /tmp/pjproject-2.10.tar.bz2
[pjproject]  Verify successful
[pjproject]  Verifying /tmp/pjproject-2.10.tar.bz2
[pjproject]  Verify successful
[pjproject]  Unpacking /tmp/pjproject-2.10.tar.bz2
[pjproject]  Applying patches 
/home/asterisk/rpmbuild/BUILD/asterisk-16.13.0-rc2/third-party/pjproject/patches 
/home/asterisk/rpmbuild/BUILD/asterisk-16.13.0-rc2/third-party/pjproject/source

[pjproject]  Applying user.mak
[pjproject]  Applying custom include file patches/config_site.h
[pjproject]  Applying custom include file patches/asterisk_malloc_debug.h
[pjproject]  Rebuilding
[pjproject]  Configuring with --build=x86_64-redhat-linux-gnu 
--host=x86_64-redhat-linux-gnu --prefix=/opt/pjproject 
--disable-speex-codec --disable-speex-aec --disable-bcg729 
--disable-gsm-codec --disable-ilbc-codec --disable-l16-codec 
--disable-g722-codec --disable-g7221-codec --disable-opencore-amr 
--disable-silk --disable-opus --disable-video --disable-v4l2 
--disable-sound --disable-ext-sound --disable-sdl --disable-libyuv 
--disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc 
--without-external-pa --without-external-srtp --disable-resample 
--disable-g711-codec --enable-epoll

checking for bundled pjproject... yes
..
checking for bridges/bridge_softmix/include/hrirs.h... yes
checking for mandatory modules:  PJPROJECT GSM ILBC IMAP_TK LIBEDIT 
SRTP... ok

configure: creating ./config.status
config.status: creating build_tools/menuselect-deps
config.status: creating makeopts
config.status: creating include/asterisk/autoconfig.h

And then asterisk builds pjproject :

..
make -C 
/home/asterisk/rpmbuild/BUILD/asterisk-16.13.0-rc2/third-party/pjproject/source/pjlib//build 
libpj-x86_64-redhat-linux-gnu.a
make -f 
/home/asterisk/rpmbuild/BUILD/asterisk-16.13.0-rc2/third-party/pjproject/source/build/rules.mak 
APP=PJLIB app=pjlib ../lib/libpj-x86_64-redhat-linux-gnu.a
make[4]: Entering directory 
'/home/asterisk/rpmbuild/BUILD/asterisk-16.13.0-rc2/third-party/pjproject/source/pjlib/build'

...

Very puzzled.

sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts

2020-09-05 Thread sean darcy

asterisk-16.13.0-rc2. Fedora 32

pjsip won't load because of undefined symbols:

[Sep  4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error 
loading module 'func_pjsip_aor.so': 
/usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: 
ast_sip_location_retrieve_aor_contacts
[Sep  4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error 
loading module 'res_pjsip_dlg_options.so': 
/usr/lib64/asterisk/modules/res_pjsip_dlg_options.so: undefined symbol: 
ast_sip_add_header
[Sep  4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error 
loading module 'res_pjsip_transport_websocket.so': 
/usr/lib64/asterisk/modules/res_pjsip_transport_websocket.so: undefined 
symbol: ast_sip_create_serializer
[Sep  4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error 
loading module 'func_pjsip_contact.so': 
/usr/lib64/asterisk/modules/func_pjsip_contact.so: undefined symbol: 
ast_sip_get_contact_status_label
[Sep  4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error 
loading module 'chan_pjsip.so': 
/usr/lib64/asterisk/modules/chan_pjsip.so: undefined symbol: 
ast_sip_cli_traverse_objects


and so on.

module show like res_pjproject
Module Description 
Use Count  Status  Support Level
res_pjproject.so   PJPROJECT Log and Utility Support 
1  Running  core

1 modules loaded

module load res_pjsip
Unable to load module res_pjsip
Command 'module load res_pjsip' failed.
ERROR[141535]: loader.c:281 module_load_error: Error loading module 
'res_pjsip': /usr/lib64/asterisk/modules/res_pjsip.so: undefined symbol: 
ast_statsd_log_full_va


module load chan_pjsip
Unable to load module chan_pjsip
Command 'module load chan_pjsip' failed.
ERROR[141780]: loader.c:281 module_load_error: Error loading module 
'chan_pjsip': /usr/lib64/asterisk/modules/chan_pjsip.so: undefined 
symbol: ast_sip_cli_traverse_objects


/usr/include/asterisk exists, with all the .h files, owned by root, 
permissions 644. For instance:


grep ast_sip_cli_traverse_objects /usr/include/asterisk/*
...
/usr/include/asterisk/res_pjsip_cli.h:char 
*ast_sip_cli_traverse_objects(struct ast_cli_entry *e, int cmd, struct 
ast_cli_args *a);



Do I need to preload some module?

Any help appreciated.


--
_
-- 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] permission woes on systemd

2020-01-22 Thread sean darcy

On 1/22/20 11:51 AM, Michael L. Young wrote:


- Original Message -

From: "sean darcy" 
To: "Asterisk Users Mailing List, Non-Commercial Discussion" 

Sent: Tuesday, January 21, 2020 9:22:28 PM
Subject: [asterisk-users] permission woes on systemd


[..]


So why would starting asterisk as user asterisk work, but fail using
systemd ?



Have you checked SELinux?  After creating the configuration files, did you run 
'restorecon' on the appropriate asterisk directories?  If not, the files are 
not labeled correctly and SELinux might be denying access.

Just a thought.

Michael

(elguero)



Yup. That was it.

Thanks.


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] permission woes on systemd

2020-01-21 Thread sean darcy
I'm running asterisk 16 on Fedora 31. If I start asterisk as user 
asterisk, all goes well. But if I start asterisk from systemd:


asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: res_sorcery_config.c:320 
sorcery_config_internal_load: Unable to load config file 'pjsip.conf'
Jan 21 19:36:47 asterisk.riverside asterisk[1411]: [Jan 21 19:36:47] 
ERROR[1411]: config_options.c:710 aco_process_config: Unable to load 
config file 'confbridge.conf'
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: app_confbridge.c:4351 
load_module: Unable to load config. Not loading module.
asterisk[1411]: [Jan 21 19:36:47] WARNING[1411]: loader.c:2381 
load_modules: Some non-required modules failed to load.
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 
load_modules: res_stun_monitor declined to load.
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 
load_modules: res_xmpp declined to load.
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 
load_modules: Declined modules which depend on res_xmpp: chan_motif
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 
load_modules: chan_iax2 declined to load.
asterisk[1411]: [Jan 21 19:36:47] ERROR[1411]: loader.c:2396 
load_modules: app_confbridge declined to load.


It seems to be a permission problem:

module load res_xmpp.so
Unable to load module res_xmpp.so
Command 'module load res_xmpp.so' failed.
  == Parsing '/etc/asterisk/xmpp.conf': Not found (Permission denied)

module load res_stun_monitor.so
Unable to load module res_stun_monitor.so
Command 'module load res_stun_monitor.so' failed.
  == Parsing '/etc/asterisk/res_stun_monitor.conf': Not found 
(Permission denied)


module load chan_iax2.so
Unable to load module chan_iax2.so
Command 'module load chan_iax2.so' failed.
  == Parsing '/etc/asterisk/iax.conf': Not found (Permission denied)

But the files are all there, with appropriate permissions (I think) :

ls -l /etc/asterisk | grep 'xmpp\|monitor\|iax\|confbridge'
-rw-r--r--. 1 asterisk asterisk 23674 Dec 12 19:54 confbridge.conf
-rw-r--r--. 1 asterisk asterisk  1250 Dec 12 19:54 iax.conf
-rw-r--r--. 1 asterisk asterisk  2401 Dec 19 18:30 iaxprov.conf
-rw-r--r--. 1 asterisk asterisk  1403 Dec 12 19:54 res_stun_monitor.conf
-rw-r--r--. 1 asterisk asterisk  2728 Dec 12 19:55 xmpp.conf

cat /usr/lib/systemd/system/asterisk.service | grep -v '#'
[Unit]
Description=Asterisk PBX and telephony daemon.
After=nss-lookup.target

[Service]
Type=simple
Environment=HOME=/var/lib/asterisk
WorkingDirectory=/var/lib/asterisk
User=asterisk
Group=asterisk
ExecStart=/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf
ExecStop=/usr/sbin/asterisk -rx 'core stop now'
ExecReload=/usr/sbin/asterisk -rx 'core reload'

PrivateTmp=true

[Install]
WantedBy=multi-user.target

So why would starting asterisk as user asterisk work, but fail using 
systemd ?


Any help appreciated.

sean




--
_
-- 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] USB dahdi fxo ?

2019-12-14 Thread sean darcy

On 12/14/19 11:29 AM, Greg Troxel wrote:

sean darcy  writes:


There is also the ObiHai OBi202 with an OBiLine, which provides an FXO
port remoted over SIP.  (I am not sure if this is discontinued.)


"FXO port remoted over SIP"?

I have an analog phone system. I can use the obi202 to connect the
system to asterisk ? That is, the 202 will connect a outgoing call
from one of its phone ports to asterisk, and connect an incoming call
from asterisk to that same phone port ?


The obi202 is sort of a mini pbx itself.  It has 4 slots for programming
SIP service providers (ITSP and SP, split, but four of each).  Then
there are call routing strings you can program to tell it what to do.

On mine, I have three Service Providers configured for SIP, all with
ITSP profile A.  Basically, PH (first FXS) has a sip login to asterisk
via SP2, and a "primary service" of the SP2.  So while I can dial **N to
make it do other things (an obi thing), if I pick it up and dial my
hello world extension it works.  PH2 is similar but on SP3.

Then, I have another sip login on SP1, for the FXO port, and an inbound
call route to sp1(NN) - just to have a dialplan destination for
incoming calls.  When the POTS line rings, asterisk gets an INVITE and
can route it back out to PH, or whatever.  I have set the 'spoof
callerid" flag on the obi, so that the callerid from POTS is passed to
asterisk as the origin.

On SP1, I have inbound call route set to "li", so that calls from
asterisk go out the POTS line.

I asked on the list earlier, I think, about the wisdom of separate SIP
logins to logically separate these, vs trying to mulitplex based on
digitstrings.  I concluded that especially given that I had no need for
more service providers in the obi, that I might as well assign SP1/2/3
to LI/PH/PH2, and be able to tell the obi "calls arriving on SP3 just go
to ph2" rather than having to sort inbound calls by destination (which I
suspect is doable).  There is more complexity already inherent in the
system than seems fun, so I am tending to reduce it when it doesn't cost
me funtionality.

I have encountered the dreaded spurious touch tone problem, in a way
where they keep going.  I believe this is due to confusion about
signaling mode and talk-off, and have set the obi explicitly to RFC2833.
That seems to work, but I'm not sure.


BTW, the 202 is one sale today at Amazon !


Keep in mind that the 202 provides two FXS ports.  You need the OBiLINE
also (plugs into USB port on 202) to get an FXO port.The FXS 



This is spectacular. Thanks for all the info.

But the dreaded FXO/FXS issue. It's like trying to remember linear algebra.

If I'm plugging the line from analog phone system into the 202, which 
then routes it to asterisk, I'm plugging the line into an FXS port , 
correct?


The line from the phone company is plugged into the FXO port. And since 
I'm not connecting the 202 to the phone company , I don't need one. Correct?


Sigh.

sean



--
_
-- 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] USB dahdi fxo ?

2019-12-14 Thread sean darcy

On 12/13/19 9:28 PM, Greg Troxel wrote:

sean darcy  writes:


I'm moving asterisk to a laptop, so can't use the dahdi board. Is
there any supported USB dahdi device ? I see the Sangoma USBfxo
device, but the dahdi driver no longer supports it. Anything else ?


There is also the ObiHai OBi202 with an OBiLine, which provides an FXO
port remoted over SIP.  (I am not sure if this is discontinued.)



"FXO port remoted over SIP"?

I have an analog phone system. I can use the obi202 to connect the 
system to asterisk ? That is, the 202 will connect a outgoing call from 
one of its phone ports to asterisk, and connect an incoming call from 
asterisk to that same phone port ?


BTW, the 202 is one sale today at Amazon !

sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] USB dahdi fxo ?

2019-12-13 Thread sean darcy
I'm moving asterisk to a laptop, so can't use the dahdi board. Is there 
any supported USB dahdi device ? I see the Sangoma USBfxo device, but 
the dahdi driver no longer supports it. Anything else ?


sean


--
_
-- 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] trouble building dahdi on kernel 5.2.7

2019-08-14 Thread sean darcy

On 8/14/19 6:00 PM, sean darcy wrote:

dahdi built fine on 5.1.20, but on 5.2.7:

.
   CC [M] 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o 

   SHIPPED 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o 

   LD [M] 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.o 


   Building modules, stage 2.
   MODPOST 15 modules
ERROR: "vpmadtreg_register" 
[/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.ko] 
undefined!
ERROR: "vpmadtreg_unregister" 
[/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.ko] 
undefined!

make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1
make[1]: *** [Makefile:1605: modules] Error 2
make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64'
make: *** [Makefile:74: modules] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep)

Any ideas ?

sean



And yes, kernel-devel is installled.

kernel-5.1.20-200.fc29.x86_64
kernel-5.1.21-200.fc29.x86_64
kernel-5.2.7-100.fc29.x86_64
kernel-core-5.1.20-200.fc29.x86_64
kernel-core-5.1.21-200.fc29.x86_64
kernel-core-5.2.7-100.fc29.x86_64
kernel-devel-5.1.20-200.fc29.x86_64
kernel-devel-5.1.21-200.fc29.x86_64
kernel-devel-5.2.7-100.fc29.x86_64
kernel-headers-5.2.7-100.fc29.x86_64
kernel-modules-5.1.20-200.fc29.x86_64
kernel-modules-5.1.21-200.fc29.x86_64
kernel-modules-5.2.7-100.fc29.x86_64
kernel-tools-5.2.7-100.fc29.x86_64
kernel-tools-libs-5.2.7-100.fc29.x86_64

The same kernel packages as the 5.1 kernels.

sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] trouble building dahdi on kernel 5.2.7

2019-08-14 Thread sean darcy

dahdi built fine on 5.1.20, but on 5.2.7:

.
  CC [M] 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o
  SHIPPED 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o
  LD [M] 
/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.o

  Building modules, stage 2.
  MODPOST 15 modules
ERROR: "vpmadtreg_register" 
[/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.ko] 
undefined!
ERROR: "vpmadtreg_unregister" 
[/home/asterisk/rpmbuild/BUILD/linux-dade6ac/drivers/dahdi/dahdi_vpmadt032_loader.ko] 
undefined!

make[2]: *** [scripts/Makefile.modpost:91: __modpost] Error 1
make[1]: *** [Makefile:1605: modules] Error 2
make[1]: Leaving directory '/usr/src/kernels/5.2.7-100.fc29.x86_64'
make: *** [Makefile:74: modules] Error 2
error: Bad exit status from /var/tmp/rpm-tmp.F8F4dL (%prep)

Any ideas ?

sean


--
_
-- 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] pjsip endoint woes

2019-04-09 Thread sean darcy

On 4/9/19 12:14 PM, George Joseph wrote:



On Tue, Apr 9, 2019 at 9:28 AM sean darcy <mailto:seandar...@gmail.com>> wrote:


On 4/8/19 6:18 AM, Joshua C. Colp wrote:
 > On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:
 >> On 4/5/19 10:36 AM, sean darcy wrote:
 >>> I'm trying to set up pjsip to work with an obi202 and google
voice. But
 >>> I can't configure the endpoint.
 >>>
 >>> pjsip:
 >>>
 >>> [obi202-auth](!)
 >>> type = auth
 >>> auth_type = userpass
 >>> password = 
 >>>
 >>> [obi202-aor](!)
 >>> type = aor
 >>> max_contacts = 2
 >>>
 >>> ; = endpoints  
 >>>
 >>> [gv-voice](obi202-endpoint)
 >>> auth = gv-voice
 >>> aors = gv-voice
 >>> identify_by=auth_username
 >>> ;identify_by=username ; I also tried this. Same result.
 >>> context = gv-voice
 >>>
 >>> [gv-voice](obi202-auth)
 >>> username = gv-voice
 >>>
 >>> [gv-voice](obi202-aor)
 >>>




 >>> Any help appreciated.
 >>>
 >>> sean
 >>>
 >>>
 >>
 >> I'm expecting gv-voice to be the "matching endpoint". The INVITE has
 >> gv-voice as the "Contact:" . Isn't this the "Username" in pjsip
"auth" ?
 >
 > Nope. The Contact is never considered for that. The From username
is what is matched for an endpoint using the "username" option. The
authentication username is what is matched for an endpoint using the
"auth_username" option but you also need to ensure it is enabled in
"endpoint_identifier_order" global option.

Thanks for the reply.

auth_username seems to be enabled:

asterisk*CLI> pjsip show identifiers
Identifier Names:
name not specified
ip
username
anonymous
header
auth_username

Is the order a problem ?

I set:

endpoint_identifier_order=auth_username,"name not
specified",ip,username,anonymous,header

restarted. No errors.

But no effect on the identifier order.

  From obi202 :

SIP Credentials
Parameter Name  Value
AuthUserName    gv-voice
AuthPassword    password

 >This also requires the endpoint to actually authenticate.

Not sure what this means. Of course, I agree, but how do I make this
happen?

BTW, this is 16.3.0.

sean


So you're using the obi with 2 itsp accounts, one Google Voice and the 
other Asterisk yeah?  Then forwarding calls between them?
Are the credentials defined in the gv-voice auth object also configured 
in the obi account that points to Asterisk?  Is the obi registering 
successfully to Asterisk?


In this situation it might be better to treat the OBi like an ITSP in 
Asterisk.  Instead of using registration and userid/password 
authentication, add an "identify" object that matches on the obi's ip 
address and points to the gv-voice endpoint...


[gv-voice]
type = identify
match = /
endpoint = gv-voice

This way you don't need to set up any auth objects at all and you just 
need to set "identify = ip" on the endpoint.




The obi may not be at the same ip address. I use it when I travel.

But this was solved. It was the identifier order.

2 points for anyone with this issue:

"endpoint_identifier_order=" must be in a [global] section, and needs 
"type = global" .


Also, pjsip reload won't make the change, you need to restart.

WFM now !

Thanks for all the help.

sean




--
_
-- 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] pjsip endoint woes

2019-04-09 Thread sean darcy

On 4/8/19 6:18 AM, Joshua C. Colp wrote:

On Sat, Apr 6, 2019, at 10:04 AM, sean darcy wrote:

On 4/5/19 10:36 AM, sean darcy wrote:

I'm trying to set up pjsip to work with an obi202 and google voice. But
I can't configure the endpoint.

pjsip:

[obi202-auth](!)
type = auth
auth_type = userpass
password = 

[obi202-aor](!)
type = aor
max_contacts = 2

; = endpoints  

[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried this. Same result.
context = gv-voice

[gv-voice](obi202-auth)
username = gv-voice

[gv-voice](obi202-aor)

##

  From the pjsip logging:

<--- Received SIP request (798 bytes) from UDP::5062 --->

INVITE sip:@:5060 SIP/2.0

Call-ID: bb384ee02eab7054@10.10.11.181

Content-Length: 270

CSeq: 8001 INVITE
From: @>;tag=SP377bfeeed75f36b8e
Max-Forwards: 70
To: @>
Via: SIP/2.0/UDP :5062;branch=z9hG4bK-fec7c7c4;rport
User-Agent: OBIHAI/OBi202-3.2.2.5921
Contact: :5062>
Expires: 60
Supported: replaces
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
Content-Type: application/sdp

v=0

o=- 112746442 1 IN IP4 10.10.11.181
s=-
c=IN IP4 
t=0 0
m=audio 17076 RTP/AVP 0 101 104 8
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian

[Apr  3 13:17:12] NOTICE[1762]: res_pjsip/pjsip_distributor.c:672
log_failed_request: Request 'INVITE' from
'@10.10.11.180>' failed for ':5062' (
callid: bb384ee02eab7054@) - No matching endpoint found

Any help appreciated.

sean




I'm expecting gv-voice to be the "matching endpoint". The INVITE has
gv-voice as the "Contact:" . Isn't this the "Username" in pjsip "auth" ?


Nope. The Contact is never considered for that. The From username is what is matched for an endpoint using 
the "username" option. The authentication username is what is matched for an endpoint using the 
"auth_username" option but you also need to ensure it is enabled in 
"endpoint_identifier_order" global option.


Thanks for the reply.

auth_username seems to be enabled:

asterisk*CLI> pjsip show identifiers
Identifier Names:
name not specified
ip
username
anonymous
header
auth_username

Is the order a problem ?

I set:

endpoint_identifier_order=auth_username,"name not 
specified",ip,username,anonymous,header


restarted. No errors.

But no effect on the identifier order.

From obi202 :

SIP Credentials
Parameter Name  Value
AuthUserNamegv-voice
AuthPasswordpassword

This also requires the endpoint to actually authenticate. 


Not sure what this means. Of course, I agree, but how do I make this happen?

BTW, this is 16.3.0.

sean



--
_
-- 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] pjsip endoint woes

2019-04-06 Thread sean darcy

On 4/5/19 10:36 AM, sean darcy wrote:
I'm trying to set up pjsip to work with an obi202 and google voice. But 
I can't configure the endpoint.


pjsip:

[obi202-auth](!)
type = auth
auth_type = userpass
password = 

[obi202-aor](!)
type = aor
max_contacts = 2

; = endpoints  

[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried this. Same result.
context = gv-voice

[gv-voice](obi202-auth)
username = gv-voice

[gv-voice](obi202-aor)

##

 From the pjsip logging:

<--- Received SIP request (798 bytes) from UDP::5062 --->

INVITE sip:@:5060 SIP/2.0

Call-ID: bb384ee02eab7054@10.10.11.181

Content-Length: 270

CSeq: 8001 INVITE
From: @>;tag=SP377bfeeed75f36b8e
Max-Forwards: 70
To: @>
Via: SIP/2.0/UDP :5062;branch=z9hG4bK-fec7c7c4;rport
User-Agent: OBIHAI/OBi202-3.2.2.5921
Contact: :5062>
Expires: 60
Supported: replaces
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
Content-Type: application/sdp

v=0

o=- 112746442 1 IN IP4 10.10.11.181
s=-
c=IN IP4 
t=0 0
m=audio 17076 RTP/AVP 0 101 104 8
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian

[Apr  3 13:17:12] NOTICE[1762]: res_pjsip/pjsip_distributor.c:672 
log_failed_request: Request 'INVITE' from 
'@10.10.11.180>' failed for ':5062' (

callid: bb384ee02eab7054@) - No matching endpoint found

Any help appreciated.

sean




I'm expecting gv-voice to be the "matching endpoint". The INVITE has 
gv-voice as the "Contact:" . Isn't this the "Username" in pjsip "auth" ?




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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] pjsip endoint woes

2019-04-05 Thread sean darcy
I'm trying to set up pjsip to work with an obi202 and google voice. But 
I can't configure the endpoint.


pjsip:

[obi202-auth](!)
type = auth
auth_type = userpass
password = 

[obi202-aor](!)
type = aor
max_contacts = 2

; = endpoints  

[gv-voice](obi202-endpoint)
auth = gv-voice
aors = gv-voice
identify_by=auth_username
;identify_by=username ; I also tried this. Same result.
context = gv-voice

[gv-voice](obi202-auth)
username = gv-voice

[gv-voice](obi202-aor)

##

From the pjsip logging:

<--- Received SIP request (798 bytes) from UDP::5062 ---> 



INVITE sip:@:5060 SIP/2.0 



Call-ID: bb384ee02eab7054@10.10.11.181 



Content-Length: 270 



CSeq: 8001 INVITE
From: @>;tag=SP377bfeeed75f36b8e
Max-Forwards: 70
To: @>
Via: SIP/2.0/UDP :5062;branch=z9hG4bK-fec7c7c4;rport
User-Agent: OBIHAI/OBi202-3.2.2.5921
Contact: :5062>
Expires: 60
Supported: replaces
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
Content-Type: application/sdp

v=0 



o=- 112746442 1 IN IP4 10.10.11.181
s=-
c=IN IP4 
t=0 0
m=audio 17076 RTP/AVP 0 101 104 8
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:8 PCMA/8000
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian

[Apr  3 13:17:12] NOTICE[1762]: res_pjsip/pjsip_distributor.c:672 
log_failed_request: Request 'INVITE' from 
'@10.10.11.180>' failed for ':5062' (

callid: bb384ee02eab7054@) - No matching endpoint found

Any help appreciated.

sean





--
_
-- 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] why doesn't extension "s" work ?

2019-03-29 Thread sean darcy

On 3/29/19 10:14 AM, Eric Wieling wrote:
Think of "s" as meaning "stupid" because calls from devices too stupid 
to send the dialed number are routed to the "s" extension.


Any incoming calls which includes the dialed number would NOT be sent to 
extension "s", those calls will match whatever the dialed number is.


On 03/28/2019 08:32 PM, sean darcy wrote:

I'm using "s" extension in my dialplan:

[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or 
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)})  ; 
PJSIP_HEADER(read,To)

    same=>n,

But when a call comes in to the gv-voice context, "s" doesn't match 
the extension:


res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' 
(UDP:10.10.10.80:5062) to extension '' rejected because 
extension not found in context 'gv-voice'.


I thought "s" (as in start ?) would match any extension sent to that 
context.


sean




OK. Thanks to both of you. I'll use _X. and I'll remember "s" as in 
"stupid" !


sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] why doesn't extension "s" work ?

2019-03-28 Thread sean darcy

I'm using "s" extension in my dialplan:

[gv-voice]
exten => s,1,Verbose(callerid is "${CALLERID(all)}" or 
"${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)})  ; PJSIP_HEADER(read,To)

   same=>n,

But when a call comes in to the gv-voice context, "s" doesn't match the 
extension:


res_pjsip_session.c:2991 new_invite: Call from 'gv-voice' 
(UDP:10.10.10.80:5062) to extension '' rejected because 
extension not found in context 'gv-voice'.


I thought "s" (as in start ?) would match any extension sent to that 
context.


sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] pattern matching "+"

2019-03-15 Thread sean darcy

From my provider I get extensions of:

+1<10digit number>
1<10 digit number>
<10 digit number>

seemingly randomly.

What I'd like to do is

exten=_!1234567890,1,Answer()

which would match anything ending in 1234567890.

But that doesn't work since ! can only be used at the end of a pattern.

I tried

[+\ ][1\ ]1234567890

which didn't work, probably because "\ " means  space, not nothing.

Any suggestions?

sean


--
_
-- 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] trouble removing + sign

2019-02-14 Thread sean darcy

On 2/14/19 4:23 AM, Administrator TOOTAI wrote:

Le 14/02/2019 à 00:12, sean darcy a écrit :
I'm using BLACKLIST() to check numbers, which does not like leading + 
signs. I want to test if there is a plus sign, and then remove it.


I tried:

  ;  strip leading plus sign
   same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
   same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) 
= ${CALLERID(num):1})

   same=>n,GotoIf(${BLACKLIST()}?make-em-wait)

but it's stripping the first character + sign or not. The callerid is 
1203XX


 -- Executing [s@hangup-spam:3] Verbose("PJSIP/2667075-000b", 
" callerid 0:1 is 1 ") in new stack

  callerid 0:1 is 1
 -- Executing [s@hangup-spam:4] ExecIf("PJSIP/2667075-000b", 
"0?Set(CALLERID(num) = 203XXX") in new stack
 -- Executing [s@hangup-spam:5] GotoIf("PJSIP/2667075-000b", 
"0?make-em-wait") in new stack


ExecIf correctly finds the comparison false(the "0"), but still 
executes the appiftrue .


What am I missing ?


Try ExecIf($["x${CALLERID(num):0:1}" == "x+"]?Set(CALLERID(num) = 
${CALLERID(num):1})


Or you could use somethjing like

exten = _X.,1,NoOp(Your dialplan)
  same = n,...
exten = _+.,1,Goto(${EXTEN:1},1)



I like using the "x" before caller id. That deals with caller id null 
values.


I also agree that the Set equal sign should not have spaces on either side.

But, the problem here was : no closing parens for the ExecIf !

Thanks for the help. I figured this out because I kept changing the line 
based on your suggestions.


sean


--
_
-- 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] trouble removing + sign

2019-02-13 Thread sean darcy

On 2/13/19 6:22 PM, Dovid Bender wrote:

Try == in your gotoif (instead of =)




Regards,

Dovid



  Original Message



From: seandar...@gmail.com
Sent: February 14, 2019 01:14
To: asterisk-users@lists.digium.com
Reply-to: asterisk-users@lists.digium.com
Subject: [asterisk-users] trouble removing + sign


I'm using BLACKLIST() to check numbers, which does not like leading +
signs. I want to test if there is a plus sign, and then remove it.

I tried:

   ;  strip leading plus sign
    same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
    same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) =
${CALLERID(num):1})
    same=>n,GotoIf(${BLACKLIST()}?make-em-wait)

but it's stripping the first character + sign or not. The callerid is
1203XX

  -- Executing [s@hangup-spam:3] Verbose("PJSIP/2667075-000b", "
callerid 0:1 is 1 ") in new stack
   callerid 0:1 is 1
  -- Executing [s@hangup-spam:4] ExecIf("PJSIP/2667075-000b",
"0?Set(CALLERID(num) = 203XXX") in new stack
  -- Executing [s@hangup-spam:5] GotoIf("PJSIP/2667075-000b",
"0?make-em-wait") in new stack

ExecIf correctly finds the comparison false(the "0"), but still executes
the appiftrue .

What am I missing ?


--


Tried the double equal sign. Same result.

sean



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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] trouble removing + sign

2019-02-13 Thread sean darcy
I'm using BLACKLIST() to check numbers, which does not like leading + 
signs. I want to test if there is a plus sign, and then remove it.


I tried:

 ;  strip leading plus sign
  same => n, Verbose( callerid 0:1 is ${CALLERID(num):0:1} )
  same => n,ExecIf($["${CALLERID(num):0:1}" = "+"]?Set(CALLERID(num) = 
${CALLERID(num):1})

  same=>n,GotoIf(${BLACKLIST()}?make-em-wait)

but it's stripping the first character + sign or not. The callerid is 
1203XX


-- Executing [s@hangup-spam:3] Verbose("PJSIP/2667075-000b", " 
callerid 0:1 is 1 ") in new stack

 callerid 0:1 is 1
-- Executing [s@hangup-spam:4] ExecIf("PJSIP/2667075-000b", 
"0?Set(CALLERID(num) = 203XXX") in new stack
-- Executing [s@hangup-spam:5] GotoIf("PJSIP/2667075-000b", 
"0?make-em-wait") in new stack


ExecIf correctly finds the comparison false(the "0"), but still executes 
the appiftrue .


What am I missing ?


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] what service does asterisk need to avoid netsock error ?

2019-01-15 Thread sean darcy
I'm running Fedora 29. asterisk starts with a systemd service at boot. 
On any reboot I get a LOT of :


[Jan 15 09:30:26] ERROR[1162]: netsock2.c:541 ast_sockaddr_hash: Unknown 
address family '0'.
[Jan 15 09:30:35] ERROR[1161]: netsock2.c:541 ast_sockaddr_hash: Unknown 
address family '0'.
[Jan 15 09:30:45] ERROR[1155]: netsock2.c:541 ast_sockaddr_hash: Unknown 
address family '0'.
[Jan 15 09:30:45] ERROR[1155]: netsock2.c:541 ast_sockaddr_hash: Unknown 
address family '0'.


If I "systemctl restart asterisk", no more errors.

I think this means something needs to start before asterisk starts. My 
asterisk.service has:


 [Unit]
Description=Asterisk PBX and telephony daemon.
After=network.target

but obviously that's not enough. What else needs to start before asterisk ?


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Surprise: DAHDI 3.0.0. Analog TDM cards EOL ?

2018-12-05 Thread sean darcy
I must have missed the memo, but I was surprised to see a new DAHDI 
release in downloads. Was there an announcement ? Is there a Changelog ?


Also, it seems there's no longer a wctdm module. What's the plan for the 
analog TDM cards ? Or is there one ?


sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] continuous netsock errors

2018-12-02 Thread sean darcy

I get continuous errors about "unknown address family" :

ERROR[1233]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'.
ERROR[1233]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'.
ERROR[1226]: netsock2.c:541 ast_sockaddr_hash: Unknown address family '0'.
.

If I restart asterisk, they go away.

I don't think they're any harm other than spamming the console. Is there 
any way avoid the error without a full restart ?


13.23.0

sean



--
_
-- 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] Is there any way to pass caller id to

2018-10-16 Thread sean darcy

On 10/16/18 1:42 PM, Antony Stone wrote:

On Tuesday 16 October 2018 at 19:04:42, Ivan Demkovitch wrote:


Thanks all,
I did contact Callcentric about it and their tech support helped meget
those headers established. They even helped to troubleshoot Asterisk
dialplan. A the end all works as it should.


For the benefit of others who may run into the same sort of problem:

1. What did Call Centric's tech support people do?

2. What did they advise you to change?

3. What did you end up with as a working dialplan (at least, the part that
dials out to Call Centric)?

Other carriers may well work the same way as Call Centric, so this information
could be helpful to other people on similar connections.


Regards,


Antony.


+1


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Convert SIP to PJSIP

2018-09-26 Thread sean darcy

On 9/24/18 2:57 PM, John T. Bittner wrote:

Hello all,

I am having some trouble converting this setup from SIP to PJSIP. Any 
help is much appreciated.


I used the converter script and get most of it but don’t see a 
registration entry.


How do you convert this entry into PJSIP.

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321 



[17185553321]

type = peer

host = sip.ringcentral.com

transport=udp

defaultuser=332940285773   ; Authentication username for outbound 
proxies


username = 332940285773

fromuser=17185553321   ; Many SIP providers require this

fromdomain=sip.ringcentral.com

secret = ARi4uYb2Mz

canreinvite = no

disallow = all

allow = ulaw

nat = yes

dtmfmode = auto

rfc2833compensate = yes

trustrpid = yes

usereqphone = yes  ; This provider requires ";user=phone" on URI

callcounter = yes  ; Enable call counter for parallel 
outbound calls


busylevel = 2  ; Signal busy at 2 or more calls (feel 
free to adjust)


outboundproxy=sip12.ringcentral.com:5090

This is what it was converted too: But nothing for the registration ?

[17185553321]

type = aor

contact = sip:332940285...@sip.ringcentral.com

[17185553321]

type = identify

endpoint = 17185553321

match = sip.ringcentral.com

[17185553321]

type = auth

username = 17185553321

password = ARi4uYb2Mz

[17185553321]

type = endpoint

dtmf_mode = none

disallow = all

allow = ulaw

rtp_symmetric = yes

rewrite_contact = yes

outbound_proxy = sip12.ringcentral.com:5090

direct_media = no

from_user = 17185553321

from_domain = sip.ringcentral.com

device_state_busy_at = 2

auth = 17185553321

outbound_auth = 17185553321

aors = 17185553321



I'd try the convert script again and make sure the input file is 
sip.conf. A lot of this pjsip config doesn't make sense.


And I hope these numbers and passwords are fake !



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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Convert SIP to PJSIP

2018-09-26 Thread sean darcy

On 9/24/18 5:04 PM, John T. Bittner wrote:

Hello all,

I am having some trouble getting this to work under pjsip. Any help is 
much appreciated.


I used the converter script and I see it register but can’t receive or 
send to ringcentral.


Anyone get this working with PJSIP?

Works with chan_sip…

This working sip config.

register => 
17185553...@sip.ringcentral.com:ARi4uYb2Mz:332940285...@sip12.ringcentral.com:5090/17185553321 



[17185553321]

type = peer

host = sip.ringcentral.com

transport=udp

defaultuser=332940285773   ; Authentication username for outbound 
proxies


username = 332940285773

fromuser=17185553321   ; Many SIP providers require this

fromdomain=sip.ringcentral.com

secret = ARi4uYb2Mz

canreinvite = no

disallow = all

allow = ulaw

nat = yes

dtmfmode = auto

rfc2833compensate = yes

trustrpid = yes

usereqphone = yes  ; This provider requires ";user=phone" on URI

callcounter = yes  ; Enable call counter for parallel 
outbound calls


busylevel = 2  ; Signal busy at 2 or more calls (feel 
free to adjust)


outboundproxy=sip12.ringcentral.com:5090



What's your pjsip config ?



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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] hangup the _called_ channel ?

2018-09-12 Thread sean darcy

On 9/12/18 1:32 PM, Joshua Colp wrote:

On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:

On 9/12/18 1:22 PM, Joshua Colp wrote:

On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:

I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.

SIP/mycall-x calls and bridges with DAHDI/1-1.

I send SIP/  to listen to a long, very long, file.


Define "send". How are you doing it?


GoSub(play-long-file,s,1)


You can't have a channel both in dialplan directly and also bridged to another 
channel at the same time. There's not enough context or information to really 
be able to answer without understanding fully.



Maybe this will help explain it. Here's the cli:

Executing [s@incoming:7] Dial("SIP/incall-0001", 
"DAHDI/g0,55,tTD(:1)") in new stack

-- Called DAHDI/g0
-- DAHDI/1-1 answered SIP/incall-0001
-- Channel DAHDI/1-1 joined 'simple_bridge' basic-bridge 
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel SIP/incall-0001 joined 'simple_bridge' basic-bridge 
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>

-- SIP/incall-0001 Internal Gosub(long-file,s,1) start
-- Executing [s@long-file:1] Playback("SIP/incall-0001", 
"long-file") in new stack
--  Playing 'long-file.slin' (language 
'en')
-- Executing [s@long-file:2] Verbose("SIP/incall-0001", 
"bridgepeer is DAHDI/1-1") in new stack

Executing [s@long-file:3] Hangup("SIP/incall-0001", "") in new stack
  == Spawn extension (long-file, s, 3) exited non-zero on 
'SIP/incall-0001'
[Sep 12 13:06:06] NOTICE[2217][C-0001]: app_stack.c:1082 gosub_run: 
SIP/callcentric20-0001 Abnormal 'Gosub(long-file,s,1)' exit. 
Popping routine return locations.
-- Channel SIP/incall left 'simple_bridge' basic-bridge 
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>
-- Channel DAHDI/1-1 left 'simple_bridge' basic-bridge 
<5312c0a8-7697-4a97-b3ff-ff0484fbaf3d>

-- Hanging up on 'DAHDI/1-1'
-- Hungup 'DAHDI/1-1'

As you can see DAHDI/1-1 is not hungup until after Playback. I want to 
hangup DAHDI/1-1 before the Playback.


Thanks,




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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] hangup the _called_ channel ?

2018-09-12 Thread sean darcy

On 9/12/18 1:25 PM, sean darcy wrote:

On 9/12/18 1:22 PM, Joshua Colp wrote:

On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:

I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.

SIP/mycall-x calls and bridges with DAHDI/1-1.

I send SIP/  to listen to a long, very long, file.


Define "send". How are you doing it?




GoSub(play-long-file,s,1)

[play-long-file]
exten=s,1,  ;;; Here I want to hangup DAHDI/1-1, the called channel
same=n,Playback(very-long-file)
same=n,Hangup()

Is there a better way ?



And I'm using dynamic features, applicationmap.

play-file=*8,peer,GoSub,"pay-long-file,s,1"




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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] hangup the _called_ channel ?

2018-09-12 Thread sean darcy

On 9/12/18 1:22 PM, Joshua Colp wrote:

On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:

I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.

SIP/mycall-x calls and bridges with DAHDI/1-1.

I send SIP/  to listen to a long, very long, file.


Define "send". How are you doing it?


GoSub(play-long-file,s,1)

[play-long-file]
exten=s,1,  ;;; Here I want to hangup DAHDI/1-1, the called channel
same=n,Playback(very-long-file)
same=n,Hangup()

Is there a better way ?


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] hangup the _called_ channel ?

2018-09-12 Thread sean darcy

On 9/12/18 1:22 PM, Joshua Colp wrote:

On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:

I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.

SIP/mycall-x calls and bridges with DAHDI/1-1.

I send SIP/  to listen to a long, very long, file.


Define "send". How are you doing it?




GoSub(play-long-file,s,1)

[play-long-file]
exten=s,1,  ;;; Here I want to hangup DAHDI/1-1, the called channel
same=n,Playback(very-long-file)
same=n,Hangup()

Is there a better way ?


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] hangup the _called_ channel ?

2018-09-12 Thread sean darcy
I understand that HangUp() hangs up the calling channel. I want to 
hangup the called channel.


SIP/mycall-x calls and bridges with DAHDI/1-1.

I send SIP/  to listen to a long, very long, file.

GoSub(play-long-file,s,1)

[play-long-file]
exten=s,1,  ;;; Here I want to hangup DAHDI/1-1, the called channel
same=n,Playback(very-long-file)
same=n,Hangup()

How do I hangup the called channel, and leave the calling channel 
listening to the file ?



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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] STUN re-evalutation every 2 minutes ??

2018-09-01 Thread sean darcy

13.21.0

Every 2-3 minutes:

Sep  1 16:00:57] WARNING[150257]: res_stun_monitor.c:140 
stun_monitor_request: STUN poll got no response. Re-evaluating STUN 
server address.
[Sep  1 16:02:18] NOTICE[150257]: res_stun_monitor.c:151 
stun_monitor_request: Old external address/port :42562 now 
seen as :33904.

 IAX, got a network change message, renewing all IAX registrations.
 SIP, got a network change message, renewing all SIP registrations.

Always just for a different port number.

I've tried a number of STUN servers with the same result. Now using 
counterpath :


/etc/asterisk/res_stun_monitor.conf:stunaddr = stun.counterpath.net

Probably harmless, but odd.

sean



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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Community forum ?

2018-08-30 Thread sean darcy
I see a lot of tag lines on posts for the Asterisk Community Forum. Is 
that forum supposed to supersede this mailing list ?


sean


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] getting invites to rtp ports ??

2018-08-30 Thread sean darcy

On 08/29/2018 09:33 PM, John Covici wrote:

OK, Thanks.  I have a couple of questions -- the line numbers do not
match exactly, so can you tell me a couple of lines before and after
the line in question?  Also, when will this be logged, if its only
during sip debug, I need to change it to log when I can see it more
readily.

Thanks.

On Wed, 29 Aug 2018 20:31:15 -0400,
sean darcy wrote:


On 08/29/2018 08:07 PM, John Covici wrote:

I wonder if I could have that patch, maybe I could add it to my
fail2ban regexp and if you have the correct regexp, I would apperciate
that as well.

Thanks.

On Wed, 29 Aug 2018 19:18:29 -0400,
Telium Support Group wrote:


Depending on log trolling (Asterisk security log) misses a lot, and also 
depends on the SIP/PJSIP folks to not change message structure (which has 
already happened numerous time).  If  you are comfortable hacking chan_sip.c 
you may prefer to get the same messages from the AMI.  It still misses a lot 
but that approach is better than nothing.

Digium warns not to use fail2ban / log trolling as a security system: 
http://forums.asterisk.org/viewtopic.php?p=159984


-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 6:33 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 11:59 AM, Telium Support Group wrote:

Block a single IP is the wrong approach (whack-a-mole).  You should consider a 
more comprehensive approach to securing your VoIP environment.  Have a look at 
this wiki:

https://www.voip-info.org/asterisk-security/



-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 09:42 AM, Carlos Rojas wrote:

Hi

Probably somebody is trying to hack your system, you should block
that ip on your firewall.

Regards

On Wed, Aug 29, 2018 at 9:34 AM, sean darcy mailto:seandar...@gmail.com>> wrote:

   I'm getting invites to very high ports every 30 seconds from a
   particular ip address:

   Retransmitting #10 (NAT) to 5.199.133.128:52734
   <http://5.199.133.128:52734>:
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
   0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
   From: mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972
   To: mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748
   Call-ID: 1504207870-295758084-609228182
   CSeq: 1 INVITE
   ...
   WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
   1504207870-295758084-609228182...

   I thought invites had to go to port 5060 or so. I don't understand
   why somebody (let's assume a bad guy) is trying ports above 5.

   sean




Ok, so the high port is not the destination port but the source port.

So I hacked the log warning in chan_sip.c on non-critical invites to show the 
source ip:

ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from
%s.\n",
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));

With that in the log, I'm now blocking the ip addresses.

Thanks,
sean


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

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

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



I agree. That's why I hacked chan_sip.c to get the addresses in the log.

I'm surprised they're not in the log by default. I must be the only person who gets these 
"non-critical invites".

sean



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

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

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

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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


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

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

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

New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started

Re: [asterisk-users] getting invites to rtp ports ??

2018-08-29 Thread sean darcy

On 08/29/2018 08:07 PM, John Covici wrote:

I wonder if I could have that patch, maybe I could add it to my
fail2ban regexp and if you have the correct regexp, I would apperciate
that as well.

Thanks.

On Wed, 29 Aug 2018 19:18:29 -0400,
Telium Support Group wrote:


Depending on log trolling (Asterisk security log) misses a lot, and also 
depends on the SIP/PJSIP folks to not change message structure (which has 
already happened numerous time).  If  you are comfortable hacking chan_sip.c 
you may prefer to get the same messages from the AMI.  It still misses a lot 
but that approach is better than nothing.

Digium warns not to use fail2ban / log trolling as a security system: 
http://forums.asterisk.org/viewtopic.php?p=159984


-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 6:33 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 11:59 AM, Telium Support Group wrote:

Block a single IP is the wrong approach (whack-a-mole).  You should consider a 
more comprehensive approach to securing your VoIP environment.  Have a look at 
this wiki:

https://www.voip-info.org/asterisk-security/



-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 09:42 AM, Carlos Rojas wrote:

Hi

Probably somebody is trying to hack your system, you should block
that ip on your firewall.

Regards

On Wed, Aug 29, 2018 at 9:34 AM, sean darcy mailto:seandar...@gmail.com>> wrote:

  I'm getting invites to very high ports every 30 seconds from a
  particular ip address:

  Retransmitting #10 (NAT) to 5.199.133.128:52734
  <http://5.199.133.128:52734>:
  SIP/2.0 401 Unauthorized
  Via: SIP/2.0/UDP
  0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
  From: mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972
  To: mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748
  Call-ID: 1504207870-295758084-609228182
  CSeq: 1 INVITE
  ...
  WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
  1504207870-295758084-609228182...

  I thought invites had to go to port 5060 or so. I don't understand
  why somebody (let's assume a bad guy) is trying ports above 5.

  sean




Ok, so the high port is not the destination port but the source port.

So I hacked the log warning in chan_sip.c on non-critical invites to show the 
source ip:

ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from
%s.\n",
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));

With that in the log, I'm now blocking the ip addresses.

Thanks,
sean


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

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

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



I agree. That's why I hacked chan_sip.c to get the addresses in the log.

I'm surprised they're not in the log by default. I must be the only person who gets these 
"non-critical invites".

sean



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

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

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

New to Asterisk? Start here:
   https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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


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

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

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

New to Asterisk? Start here:
   https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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




The patch, more accurately a hack, is in my second post above.

chan_sip.c 4127 : ast_log(LOG_WARNING, "Timeout on %s non-critic invite 
trans from %s.\n", 
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));


The added second %s shows the ip a

Re: [asterisk-users] getting invites to rtp ports ??

2018-08-29 Thread sean darcy

On 08/29/2018 11:59 AM, Telium Support Group wrote:

Block a single IP is the wrong approach (whack-a-mole).  You should consider a 
more comprehensive approach to securing your VoIP environment.  Have a look at 
this wiki:

https://www.voip-info.org/asterisk-security/



-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of sean darcy
Sent: Wednesday, August 29, 2018 10:46 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] getting invites to rtp ports ??

On 08/29/2018 09:42 AM, Carlos Rojas wrote:

Hi

Probably somebody is trying to hack your system, you should block that
ip on your firewall.

Regards

On Wed, Aug 29, 2018 at 9:34 AM, sean darcy mailto:seandar...@gmail.com>> wrote:

 I'm getting invites to very high ports every 30 seconds from a
 particular ip address:

 Retransmitting #10 (NAT) to 5.199.133.128:52734
 <http://5.199.133.128:52734>:
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP
 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
 From: mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972
 To: mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748
 Call-ID: 1504207870-295758084-609228182
 CSeq: 1 INVITE
 ...
 WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
 1504207870-295758084-609228182...

 I thought invites had to go to port 5060 or so. I don't understand
 why somebody (let's assume a bad guy) is trying ports above 5.

 sean




Ok, so the high port is not the destination port but the source port.

So I hacked the log warning in chan_sip.c on non-critical invites to show the 
source ip:

ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from %s.\n",
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));

With that in the log, I'm now blocking the ip addresses.

Thanks,
sean


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

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

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



I agree. That's why I hacked chan_sip.c to get the addresses in the log.

I'm surprised they're not in the log by default. I must be the only 
person who gets these "non-critical invites".


sean



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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] getting invites to rtp ports ??

2018-08-29 Thread sean darcy

On 08/29/2018 09:42 AM, Carlos Rojas wrote:

Hi

Probably somebody is trying to hack your system, you should block that 
ip on your firewall.


Regards

On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <mailto:seandar...@gmail.com>> wrote:


I'm getting invites to very high ports every 30 seconds from a
particular ip address:

Retransmitting #10 (NAT) to 5.199.133.128:52734
<http://5.199.133.128:52734>:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734
From: mailto:sip%3A37120116780191250@67.80.191.250>>;tag=1872048972
To: mailto:sip%3A3712011972592181418@67.80.191.250>>;tag=as3a52e748
Call-ID: 1504207870-295758084-609228182
CSeq: 1 INVITE
...
WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on
1504207870-295758084-609228182...

I thought invites had to go to port 5060 or so. I don't understand
why somebody (let's assume a bad guy) is trying ports above 5.

sean




Ok, so the high port is not the destination port but the source port.

So I hacked the log warning in chan_sip.c on non-critical invites to 
show the source ip:


ast_log(LOG_WARNING, "Timeout on %s non-critic invite trans from %s.\n", 
pkt->owner->callid,ast_sockaddr_stringify(sip_real_dst(pkt->owner)));


With that in the log, I'm now blocking the ip addresses.

Thanks,
sean


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] getting invites to rtp ports ??

2018-08-29 Thread sean darcy
I'm getting invites to very high ports every 30 seconds from a 
particular ip address:


Retransmitting #10 (NAT) to 5.199.133.128:52734:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734

From: ;tag=1872048972
To: ;tag=as3a52e748
Call-ID: 1504207870-295758084-609228182
CSeq: 1 INVITE
...
WARNING[150318]: chan_sip.c:4127 retrans_pkt: Timeout on 
1504207870-295758084-609228182...


I thought invites had to go to port 5060 or so. I don't understand why 
somebody (let's assume a bad guy) is trying ports above 5.


sean


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

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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Decoding SIP register hack

2018-05-18 Thread sean darcy

On 05/17/2018 05:29 PM, sean darcy wrote:

On 05/17/2018 04:47 PM, Daniel Tryba wrote:

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:

    WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to
register.


His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).



I'm far from a regex expert, but I don't think that last line would 
capture anything in the invite. In fact, asterisk doesn't throw any 
WARNING at all for this INVITE.


I'm not sure, but I don't even see how you can get asterisk to log these 
invites at all. There's no heading such as WARNING( or NOTICE, SECURITY, 
etc).



  That why I find it puzzling. What is he trying to do ?


There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.



Very interesting.

sen






I found these by staring at sip debug, and tying together the SIP 
retransmission id with the INVITE. That was an afternoon! Is there any 
way to automate this ? Specifically, find the INVITE that generates the 
retransmission ?


Otherwise, I can't see how anyone could block these attempts.

> There are sip servers publicly reachable that will relay INVITEs, make
> sure yours aren't.

How do I make sure my server won't relay INVITEs ?

sean


--
_
-- 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] Decoding SIP register hack

2018-05-17 Thread sean darcy

On 05/17/2018 04:47 PM, Daniel Tryba wrote:

On Thu, May 17, 2018 at 12:27:17PM -0400, sean darcy wrote:

WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to
register.


His filter doesn't only trigger on REGISTERs, see the last line of the
matches and the context for guests (which logs the pattern of the last
line of the filter on an INVITE).



I'm far from a regex expert, but I don't think that last line would 
capture anything in the invite. In fact, asterisk doesn't throw any 
WARNING at all for this INVITE.


I'm not sure, but I don't even see how you can get asterisk to log these 
invites at all. There's no heading such as WARNING( or NOTICE, SECURITY, 
etc).



  That why I find it puzzling. What is he trying to do ?


There are sip servers publicly reachable that will relay INVITEs, make
sure yours aren't. And there are only 2 kinds of operators of sip
server:
-those that have been the victim of toll fraud
-those that will be the victim of toll fraud

You can do nothing to stop this kind of traffic. The only thing you can
do is block it, either using only a whitelist (cumbersome) or generate a
blacklist with for example fail2ban or a more elaborate honeypot setup.
Or setup a proxy that will filter patterns you discover from

BTW this is not a person, this is an automated script, running most
likely on compromised machines and sending spoofed ips. These scripts
care about generating a ring on a phone (again most an abuseable/hacked
account (or purchased with CC fraud)). If they find a server that does,
it will be targetted for all kind of fraud.



Very interesting.

sen



--
_
-- 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] Decoding SIP register hack

2018-05-17 Thread sean darcy

On 05/17/2018 11:38 AM, Frank Vanoni wrote:

On Thu, 2018-05-17 at 11:18 -0400, sean darcy wrote:


3. How do I set up the server to block these ?

4. Can I stop the retransmitting of the 401 Unauthorized packets ?


I'm happy with Fail2Ban protecting my Asterisk 13. Here is my
configuration:

in /etc/asterisk/logger.conf:

messages => security,notice,warning,error


in /etc/asterisk/sip.conf:

allowguest=yes
context=unauthenticated


in /etc/asterisk/extensions.conf:

[unauthenticated]
;; Incomming calls from unauthenticated caller -> Fail2Ban
exten => _X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _X.,3,HangUp()

exten => _+X.,1,Log(WARNING,fail2ban='${CHANNEL(peerip)}')
exten => _+X.,2,Set(CDR(UserField)=SIP PEER IP: ${CHANNEL(peerip)})
exten => _+X.,3,HangUp()



in /etc/fail2ban/jail.conf:

[asterisk]
filter   = asterisk
action = iptables-allports[name=ASTERISK]
logpath  = /var/log/asterisk/messages
maxretry = 1
findtime = 86400
bantime  = 518400
enabled = true


in /etc/fail2ban/filter.d

# Fail2Ban configuration file
#
#
# $Revision: 250 $
#

[INCLUDES]

# Read common prefixes. If any customizations available -- read them
from
# common.local
#before = common.conf


[Definition]

#_daemon = asterisk

# Option:  failregex
# Notes.:  regex to match the password failures messages in the
logfile. The
#  host must be matched by a group named "host". The tag
"" can
#  be used for standard IP/hostname matching and is only an
alias for
#  (?:::f{4,6}:)?(?P\S+)
# Values:  TEXT
#
failregex = NOTICE.* .*: Registration from '.*' failed for
':.*' - Wrong password
NOTICE.* .*: Call from '.*' \((:[0-9]{1,5})?\) to
extension '.*' rejected because extension not found in context
'unauthenticated'
NOTICE.* chan_sip.c: Call from '.*' \((:[0-
9]{1,5})?\) to extension '.*' rejected because extension not found in
context 'unauthenticated'
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Username/auth name mismatch
    NOTICE.* .*: Registration from '.*' failed for
':.*' - No matching peer found
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Not a local domain
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Peer is not supposed to register
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Device does not match ACL
    NOTICE.* .*: Registration from '.*' failed for
':.*' - Device not configured to use this transport type
    NOTICE.* .*: No registration for peer '.*' \(from
\)
    NOTICE.* .*: Host  failed MD5 authentication for
'.*' \(.*\)
    NOTICE.* .*: Host  denied access to register peer
'.*'
    NOTICE.* .*: Host  did not provide proper
plaintext password for '.*'
    NOTICE.* .*: Registration of '.*' rejected: '.*' from:
''
    NOTICE.* .*: Peer '.*' is not dynamic (from )
    NOTICE.* .*: Host  denied access to register peer
'.*'
    SECURITY.* .*:
SecurityEvent="InvalidAccountID".*,Severity="Error",Service="SIP".*,Rem
oteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="FailedACL".*,Severity="Error",Service="SIP".*,RemoteAddr
ess="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="InvalidPassword".*,Severity="Error",Service="SIP".*,Remo
teAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
    SECURITY.* .*:
SecurityEvent="ChallengeResponseFailed".*,Severity="Error",Service="SIP
".*,RemoteAddress="IPV[46]/(UDP|TCP|TLS)//[0-9]+"
VERBOSE.* logger.c: -- .*IP/-.* Playing 'ss-
noservice' \(language '.*'\)
SECURITY.* .*:
SecurityEvent="ChallengeSent".*,Severity="Informational",Service="SIP".
*,AccountID="sip:.*@93.94.247.123".*,RemoteAddress="IPV[46]/(UDP|TCP|TL
S)//[0-9]+
WARNING.* .*: fail2ban=''

# Option:  ignoreregex
# Notes.:  regex to ignore. If this regex matches, the line is ignored.
# Values:  TEXT
#
ignoreregex =



Thanks. Very useful as a tutorial for fail2ban.

But I don't think it covers this SIP hack. This guy isn't trying to 
register. That why I find it puzzling. What is he trying to do ?


sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Decoding SIP register hack

2018-05-17 Thread sean darcy
I need some help understanding SIP dialog. Some actor is trying to 
access my server, but I can't figure out what he's trying to do ,or how.


I'm getting a lot of these warnings.

[May 17 10:08:08] WARNING[1532]: chan_sip.c:4068 retrans_pkt: 
Retransmission timeout reached on transmission 
_zIr9tDtBxeTVTY5F7z8kD7R.. for seqno 101


With SIP DEBUG I tracked the Call-ID to this INVITE :

<--- SIP read from UDP:192.111.139.146:29281 --->
INVITE sip:+48223079992@67.80.191.250:5060 SIP/2.0
Via: SIP/2.0/UDP 
100.149.241.68:5060;branch=z4hG4bK-966187-1---q9ft4HdLB4ZeBqs;rport=5060
Contact: 
;+sip.instance=""

Max-Forwards: 70
To: 
From: "Caller";tag=sXPNixD5Ui42V
Call-ID: _zIr9tDtBxeTVTY5F7z8kD7R..
CSeq: 101 INVITE
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
REGISTER, SUBSCRIBE, INFO

Supported: replaces
User-Agent: GSM
Allow-Events: hold, talk, conference
Accept: application/sdp
Content-Length: 771

v=0
o=CiscoSystemsSIP-IPPhone 18338 11953 IN IP4 100.149.241.68
s=SIP Call
c=IN IP4 100.149.241.68
t=0 0
m=audio 2 RTP/AVP 0 8 18 101
a=rtpmap:3 gsm/8000
a=rtpmap:96 speex/8000
a=rtpmap:97 speex/8000
a=fmtp:97 mode=2
a=rtpmap:98 speex/8000
a=fmtp:98 mode=5
a=rtpmap:99 speex/8000
a=fmtp:99 mode=7
a=rtpmap:107 speex/32000
a=fmtp:107 mode=10
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:108 ilbc/8000
a=rtpmap:113 g7231/8000
a=rtpmap:18 g729/8000
a=rtpmap:100 G726-16/8000
a=rtpmap:101 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:4 g723/8000
a=fmtp:18 annexb=no
a=rtpmap:109 ilbc/8000
a=fmtp:109 mode=20
a=rtpmap:110 telephone-event/8000
a=fmtp:110 0-15
a=ptime:20
a=sendrecv
<->
--- (15 headers 34 lines) ---
Sending to 192.111.139.146:29281 (NAT)
Sending to 192.111.139.146:29281 (NAT)
Using INVITE request as basis request - _zIr9tDtBxeTVTY5F7z8kD7R..
No matching peer for '9353' from '192.111.139.146:29281'
..
Which then generates a lot of transmissions showing Unauthorized:
..
Retransmitting #10 (NAT) to 192.111.139.146:29281:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
100.149.241.68:5060;branch=z4hG4bK-966187-1---q9ft4HdLB4ZeBqs;received=192.111.139.146;rport=29281

From: "Caller";tag=sXPNixD5Ui42V
To: ;tag=as1f60e6dd
Call-ID: _zIr9tDtBxeTVTY5F7z8kD7R..
CSeq: 101 INVITE
Server: Asterisk PBX 13.21.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk_home", 
nonce="0794806c"

Content-Length: 0


1. What's this guy trying to do ? It looks like he's trying to generate 
a call from the server to a Polish number. Why bother ?


2. What's the role of the Via and the Contact line ?  The 100.149.241.68 
seems to be a cell phone. 100.128.0.0/9 is T-mobile.


3. How do I set up the server to block these ?

4. Can I stop the retransmitting of the 401 Unauthorized packets ?

Any help appreciated.

sean


--
_
-- 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] opus from git : install questions

2018-02-06 Thread sean darcy

On 02/06/2018 07:58 AM, Tzafrir Cohen wrote:

On Sun, Feb 04, 2018 at 03:15:02PM -0500, sean darcy wrote:


On 13.9.0
https://github.com/traud/asterisk-opus

The README:

Alternatively, you can use the Makefile of this repository to create just
the shared libraries of the modules. That way, you do not have to (re-) make
your whole Asterisk.

The Makefile generates:
codecs/codec_opus_open_source.so
formats/format_ogg_opus_open_source.so
formats/format_vp8.so
res/res_format_attr_opus.so


See, e.g. the Debian package asterisk-opus:
https://packages.debian.org/source/sid/asterisk-opus

That package builds the opus modules as stand-alone modules out of the
tree of asterisk. It has a build-time dependency  on the binary package
asterisk-dev.

The list of files in it:

   /usr/lib/asterisk/modules/codec_opus_open_source.so
   /usr/lib/asterisk/modules/format_ogg_opus_open_source.so
   /usr/lib/asterisk/modules/format_vp8.so

IIRC res_format_attr_opus.so in Asterisk proper is by now good enough
and needs no patching.




Without any of the patches the asterisk build generates:

codec_opus.so

format_ogg_opus.so


Not by default.



So if I'm building outside the tree, do _not_ select opus in menuselect 
when building asterisk. Correct ? Then copy codec_opus_open_source.so, 
format_ogg_opus_open_source.so and format_vp8.so to {astmoddir} , 
/usr/lib64/asterisk/modules.




res_format_attr_opus.so

Questions:

Should the *_opus_open_source.so be in modules with the *opus.so libraries
from the asterisk build ? If not, do I build asterisk without selecting opus
? If they are in modules, how does asterisk know which library to use ?


You can either copy them to the Asterisk build directory (I do that in
another Asterisk package I maintain). The build system will just pick
them up and use them.

Alternatively, as suggested in the README above, build them outside of
the Asterisk tree (you'll need to point them to the asterisk source tree,
or at least to the installed asterisk header files and copy the
resulting .so files to the astmoddir (e.g. /usr/lib/asterisk/modules).



The res_format_attr_opus.so library ?

And another question, to enable PLC do I need to apply the patch even if I'm
installing the libraries directly ?


Which patch specifically? I'm not sure I follow.



asterisk-opus-asterisk-13.7/enable_native_plc.patch

From the README:

Out of the box, Asterisk does not detect lost (or late) RTP packets. 
Such a detection is required to conceal lost packets (PLC). PLC improves 
situations like Wi-Fi Roaming or mobile-phone handovers. This patch 
detects lost/late packets but is experimental.


patch -p1 <./asterisk-opus*/enable_native_plc.patch


Looks like this can't be built out-of-tree, and you need to patch each 
build of asterisk.







Thanks for all the work getting opus to work on asterisk.




Appreciate the quick reply.


--
_
-- 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] OAuth : xmpp.conf

2018-02-05 Thread sean darcy

On 02/03/2018 07:11 PM, sean darcy wrote:
Confused about xmpp.conf with OAuth. Let's assume I have two voice 
accounts. Are all the OAuth entries in each account ? It'd be really 
great if only separate refresh_token s were required!


For instance- painful:

[general]
...

[gv1]
.
refresh_token=gv1-token
oauth_clientid=gv1-client-id
oauth_secret=gv1-oauth-secret

[gv2]
.
refresh_token=gv2-token
oauth_clientid=gv2-client-id
oauth_secret=gv2-oauth-secret

Or just the refresh token (much better):

[general]
...
oauth_clientid=gv-client-id
oauth_secret=gv-oauth-secret

[gv1]
.
refresh_token=gv1-token

[gv2]
.
refresh_token=gv2-token

OR (just as good):

[general]
...

[oauth]!
oauth_clientid=gv-client-id
oauth_secret=gv-oauth-secret

[gv1](oauth)
.
refresh_token=gv1-token

[gv2](oauth)
.
refresh_token=gv2-token

It's painful enough to navigate the google OAuth process for just the 
refresh tokens. I hope I don't need a clientid and secret for each account!


sean



Tried it for 1 line,

[oauth](!)
oauth_clientid=gv-client-id
oauth_secret=gv-oauth-secret
.

[79](oauth)
.
refresh_token=gv1-token

Got this error:

ERROR[7954]: res_xmpp.c:3940 fetch_access_token: An error occurred while 
performing OAuth 2.0 authentication for client '.79': {

 "error": "invalid_grant",
 "error_description": "Bad Request"

Has anybody actually set this up ?

sean



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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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


[asterisk-users] opus from git : install questions

2018-02-04 Thread sean darcy


On 13.9.0
https://github.com/traud/asterisk-opus

The README:

Alternatively, you can use the Makefile of this repository to create 
just the shared libraries of the modules. That way, you do not have to 
(re-) make your whole Asterisk.


The Makefile generates:
codecs/codec_opus_open_source.so
formats/format_ogg_opus_open_source.so
formats/format_vp8.so
res/res_format_attr_opus.so

Without any of the patches the asterisk build generates:

codec_opus.so
format_ogg_opus.so

res_format_attr_opus.so

Questions:

Should the *_opus_open_source.so be in modules with the *opus.so 
libraries from the asterisk build ? If not, do I build asterisk without 
selecting opus ? If they are in modules, how does asterisk know which 
library to use ?


The res_format_attr_opus.so library ?

And another question, to enable PLC do I need to apply the patch even if 
I'm installing the libraries directly ?


Thanks for all the work getting opus to work on asterisk.

sen


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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


[asterisk-users] OAuth : xmpp.conf

2018-02-03 Thread sean darcy
Confused about xmpp.conf with OAuth. Let's assume I have two voice 
accounts. Are all the OAuth entries in each account ? It'd be really 
great if only separate refresh_token s were required!


For instance- painful:

[general]
...

[gv1]
.
refresh_token=gv1-token
oauth_clientid=gv1-client-id
oauth_secret=gv1-oauth-secret

[gv2]
.
refresh_token=gv2-token
oauth_clientid=gv2-client-id
oauth_secret=gv2-oauth-secret

Or just the refresh token (much better):

[general]
...
oauth_clientid=gv-client-id
oauth_secret=gv-oauth-secret

[gv1]
.
refresh_token=gv1-token

[gv2]
.
refresh_token=gv2-token

OR (just as good):

[general]
...

[oauth]!
oauth_clientid=gv-client-id
oauth_secret=gv-oauth-secret

[gv1](oauth)
.
refresh_token=gv1-token

[gv2](oauth)
.
refresh_token=gv2-token

It's painful enough to navigate the google OAuth process for just the 
refresh tokens. I hope I don't need a clientid and secret for each account!


sean


--
_
-- 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] SIP invite timeouts : how is someone sending invites from our server ??

2018-01-02 Thread sean darcy

On 12/30/2017 08:18 PM, Dovid Bender wrote:
Script kiddies trying to find vulnerable systems that they can make 
calls on. Lock down the box with iptables and use fail2ban to block 
them. The via is probably bogus unless a box at the DoD was comprimised.




On Sat, Dec 30, 2017 at 6:49 PM, sean darcy <seandar...@gmail.com 
<mailto:seandar...@gmail.com>> wrote:


I've been getting a lot of timeouts on non-critical invite
transactions. I turned on sip debug. They were the result of SIP
invites like this:

Retransmitting #10 (NAT) to 185.107.94.10:13057
<http://185.107.94.10:13057>:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP

215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057
From: <sip:a'or'3=3--@;transport=UDP>;tag=fptfih1e
To: <sip:00141225184741@;transport=UDP>;tag=as2913c67b
Call-ID: 5YpLDUSIs6l3xbDXsurYTu..
CSeq: 1 INVITE
Server: Asterisk PBX 13.19.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk_home",
nonce="14be1363"
Content-Length: 0

---
  WARNING[1868]: chan_sip.c:4065 retrans_pkt: Retransmission timeout
reached on transmission 5YpLDUSIs6l3xbDXsurYTu.. for seqno 1
(Non-critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
<https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions>
Packet timed out after 32000ms with no response
  WARNING[1868]: chan_sip.c:4124 retrans_pkt: Timeout on
5YpLDUSIs6l3xbDXsurYTu.. on non-critical invite transaction.

Looking up the ip addresses :

whois 185.107.94.10
.
inetnum:        185.107.94.0 - 185.107.94.255
netname:        NFORCE_ENTERTAINMENT
descr:          Serverhosting
..
organisation:   ORG-NE3-RIPE
org-name:       NForce Entertainment B.V.
org-type:       LIR
address:        Postbus 1142
address:        4700BC
address:        Roosendaal
address:        NETHERLANDS
phone: +31206919299 <tel:%2B31206919299>
...

whois 215.45.145.211
.
NetRange:       215.0.0.0 - 215.255.255.255
CIDR: 215.0.0.0/8 <http://215.0.0.0/8>
NetName:        DNIC-NET-215
NetHandle:      NET-215-0-0-0-1
Parent:          ()
NetType:        Direct Assignment
OriginAS:
Organization:   DoD Network Information Center (DNIC)
RegDate:        1998-06-04
Updated:        2011-06-21
Ref: https://whois.arin.net/rest/net/NET-215-0-0-0-1
<https://whois.arin.net/rest/net/NET-215-0-0-0-1>



OrgName:        DoD Network Information Center
OrgId:          DNIC
Address:        3990 E. Broad Street
City:           Columbus
StateProv:      OH

So how is someone on a Dutch ISP using my server to mess with a US
DoD ip address ?


-- 


I don't see how fail2ban would help. asterisk isn't rejecting anything. 
There's no attempt with username/password.


How could I use iptables to "lock it down" ? We get sip calls from all 
over. Is there something about the incoming packet we could use ? For 
instance , any packet containing a VIA instruction ? For that matter, 
can SIP be configured to drop any VIA request?


sean



--
_
-- 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] SIP invite timeouts : how is someone sending invites from our server ??

2018-01-02 Thread sean darcy

On 12/30/2017 08:10 PM, Antony Stone wrote:

On Sunday 31 December 2017 at 00:49:17, sean darcy wrote:


I've been getting a lot of timeouts on non-critical invite transactions.



So how is someone on a Dutch ISP using my server to mess with a US DoD
ip address ?


What's your setting for "allowguest" (under [general]) in
/etc/asterisk/sip.conf ?

What are your firewall rules for UDP 5060?


Antony.


allowguest=no
alwaysauthreject = yes

The only firewall rules for UDP 5060 forward the packets to asterisk.

sean


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

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

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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


[asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

2017-12-30 Thread sean darcy
I've been getting a lot of timeouts on non-critical invite transactions. 
I turned on sip debug. They were the result of SIP invites like this:


Retransmitting #10 (NAT) to 185.107.94.10:13057:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
215.45.145.211:5060;branch=z9hG4bK-524287-1---zg4cfkl50hpwpv4p;received=185.107.94.10;rport=13057

From: ;tag=fptfih1e
To: ;tag=as2913c67b
Call-ID: 5YpLDUSIs6l3xbDXsurYTu..
CSeq: 1 INVITE
Server: Asterisk PBX 13.19.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH, MESSAGE

Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk_home", 
nonce="14be1363"

Content-Length: 0

---
 WARNING[1868]: chan_sip.c:4065 retrans_pkt: Retransmission timeout 
reached on transmission 5YpLDUSIs6l3xbDXsurYTu.. for seqno 1 
(Non-critical Response) -- See 
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

Packet timed out after 32000ms with no response
 WARNING[1868]: chan_sip.c:4124 retrans_pkt: Timeout on 
5YpLDUSIs6l3xbDXsurYTu.. on non-critical invite transaction.


Looking up the ip addresses :

whois 185.107.94.10
.
inetnum:185.107.94.0 - 185.107.94.255
netname:NFORCE_ENTERTAINMENT
descr:  Serverhosting
..
organisation:   ORG-NE3-RIPE
org-name:   NForce Entertainment B.V.
org-type:   LIR
address:Postbus 1142
address:4700BC
address:Roosendaal
address:NETHERLANDS
phone:  +31206919299
...

whois 215.45.145.211
.
NetRange:   215.0.0.0 - 215.255.255.255
CIDR:   215.0.0.0/8
NetName:DNIC-NET-215
NetHandle:  NET-215-0-0-0-1
Parent:  ()
NetType:Direct Assignment
OriginAS:
Organization:   DoD Network Information Center (DNIC)
RegDate:1998-06-04
Updated:2011-06-21
Ref:https://whois.arin.net/rest/net/NET-215-0-0-0-1



OrgName:DoD Network Information Center
OrgId:  DNIC
Address:3990 E. Broad Street
City:   Columbus
StateProv:  OH

So how is someone on a Dutch ISP using my server to mess with a US DoD 
ip address ?



--
_
-- 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] How to set outgoing sip callid ?

2016-06-01 Thread sean darcy

On 05/31/2016 11:43 AM, Frank Vanoni wrote:

(CALLERID(all)="Jon Doe" <+123456789>)


So simple. just too obvious.

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


[asterisk-users] How to set outgoing sip callid ?

2016-05-31 Thread sean darcy

Calling linphone from asterisk 13.9.1.:

Dial(SIP/@sip.linphone.org)

And it works. But on the linphone side the caller is:

@ipaddress

or

2502@45.123.987.4

Is there any way to make it more descriptive, at least for the sip user 
name ? I tried setting SIPCALLID, which had no effect.


Set(SIPCALLID=Office)

Thanks,
sean


--
_
-- 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] "__sip_xmit....Success" every 15 seconds !

2016-05-12 Thread sean darcy
On 13.9. The cli log has these messages every 15 seconds. The end point 
to linphone on android.



[May 12 19:02:59] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of 
0x7effe40088b0 (len 608) to 10.10.11.95:37855 returned -2: Success
[May 12 19:03:13] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of 
0x7effe4037860 (len 608) to 10.10.11.95:37855 returned -2: Success
[May 12 19:03:27] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of 
0x7effe40088b0 (len 608) to 10.10.11.95:37855 returned -2: Success
[May 12 19:03:41] WARNING[2555]: chan_sip.c:3775 __sip_xmit: sip_xmit of 
0x7effe4037860 (len 608) to 10.10.11.95:37855 returned -2: Success



Any clue why I get I'm continually warned about Success ?

jay


--
_
-- 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] codec_opus w/ PLR and FEC for Asterisk 11

2016-04-28 Thread sean darcy
There is an opus patch for asterisk 11. 
https://github.com/seanbright/asterisk-opus/tree/asterisk-11 . But it 
doesn't have Packet Loss Resilience or Forward Error Correction, both of 
which are important for voip.



2.1.6.  Packet Loss Resilience

Audio codecs often exploit inter-frame correlations to reduce the 
bitrate at a cost in error propagation: after losing one packet, several 
packets need to be received before the decoder is able to accurately 
reconstruct the speech signal. The extent to which Opus exploits 
inter-frame dependencies can be adjusted on the fly to choose a 
trade-off between bitrate and amount of error propagation.


2.1.7.  Forward Error Correction (FEC)

Another mechanism providing robustness against packet loss is the 
in-band Forward Error Correction (FEC). Packets that are determined to 
contain perceptually important speech information, such as onsets or 
transients, are encoded again at a lower bitrate and this re-encoded 
information is added to a subsequent packet.



There is an opus patch for asterisk 13 that includes PLR and FEC. 
https://github.com/traud/asterisk-opus.


We don't have approval to move to 13. So anybody have an opus patch for 
11 that includes PLR and FEC ?


sean



--
_
-- 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] opus : patches for FEC and PLC useful ?

2016-04-05 Thread sean darcy

On 04/05/2016 04:17 AM, Tzafrir Cohen wrote:

On Mon, Apr 04, 2016 at 11:39:07AM +0200, Ludovic Gasc wrote:

We're testing this branch for a while, not with the latest commits.

For now, it works, however, time to time audio quality issues with
transcoding, but I don't know yet where is the problem.

We'll test with the latest commits.

BTW, it isn't included in the main because potential patent license issues.


Those two patches are not on the opus code and hence this is irrelevant
to them (regardless of what I personally think about the patent
concerns).

While not on the opus code itself, the patches claim to allow FEC , 
which otherwise has problems, at least with HD voice :


https://github.com/seanbright/asterisk-opus/issues/9

Or am I missing something ?

sean


--
_
-- 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] opus : patches for FEC and PLC useful ?

2016-04-03 Thread sean darcy
In a fork of seanbright's opus patch for 13 there are further patches 
for Forward Error Correction and Package Loss Concealment, both of which 
ought to very useful in voip:


https://github.com/traud/asterisk-opus

Anybody used these patches ? Puzzled why they weren't committed to the 
main patch.


sean


--
_
-- 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] registering IAX with Teliax

2016-04-02 Thread sean darcy

On 05/13/2015 03:51 PM, Greg Woods wrote:

Hopefully this is really a generic question about IAX and doesn't turn
out to be something specific to Teliax, because they haven't been too
helpful so far. All they can tell me is that my login shows "status
unknown" on their end, which prevents me from receiving inbound calls on
my Teliax number. Outbound calls through the same server work fine,
which rules out most networking issues and presumably demonstrates that
I have the correct username and password (and I checked to make sure
that my "register" declaration in iax.conf matches my dialplan for those
things). I have already configured the firewall to permit UDP port 4569
(iax).

Here is what I see in my asterisk console:



worldsys*CLI> iax2 show registry
Host  dnsmgr  UsernamePerceived Refresh
  State
63.211.239.14:4569 N   gswoods
98.245.184.191:4569 60  Registered
1 IAX2 registrations.

If I run "iax2 set debug on", here is the resulting trace. Does this
mean anything to someone who could give suggestions for debugging this?

Thanks,
--Greg


IAX2 Debugging Enabled
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00010ms  SCall: 03639  DCall: 0 [63.211.239.14:4569
]
USERNAME: gswoods
REFRESH : 60

Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
CTOKEN
Timestamp: 00010ms  SCall: 1  DCall: 03639 [63.211.239.14:4569
]
CALLTOKEN   : 51 bytes

Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ
Timestamp: 00063ms  SCall: 03639  DCall: 0 [63.211.239.14:4569
]
USERNAME: gswoods
REFRESH : 60
CALLTOKEN   : 51 bytes

Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REGAUTH
Timestamp: 00019ms  SCall: 03026  DCall: 03639 [63.211.239.14:4569
]
AUTHMETHODS : 3
CHALLENGE   : \x31\x33\x34\x35\x38\x37\x34\x39\x31
USERNAME: gswoods

Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass:
REGREQ
Timestamp: 00115ms  SCall: 03639  DCall: 03026 [63.211.239.14:4569
]
USERNAME: gswoods
REFRESH : 60
MD5 RESULT  : 74f42b964023f9caf21ad341b0f94bf7

Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass:
REGACK
Timestamp: 00997ms  SCall: 03026  DCall: 03639 [63.211.239.14:4569
]
USERNAME: gswoods
DATE TIME   : 2015-05-13  13:49:56
REFRESH : 60
APPARENT ADDRES : IPV4 98.245.184.191:4569 
CALLING NUMBER  : 3038727683

Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK
Timestamp: 00997ms  SCall: 03639  DCall: 03026 [63.211.239.14:4569
]




I believe teliax, no longer supports IAX, sadly.

sean


--
_
-- 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] PJProject Bundled Update

2016-04-01 Thread sean darcy

On 03/31/2016 11:57 AM, George Joseph wrote:

As you know, the ability to use a bundled version of pjproject was
introduced with Asterisk 13.8.0.

More info on the Asterisk Wiki

 and
in this email thread
.

Since then I've fixed a few issues related to older versions of Debian
and CentOS which you can in these 2 patches.
https://gerrit.asterisk.org//2516
https://gerrit.asterisk.org/2449

Any other feedback?  I'd like to get an idea of how many folks have
tried it.

Thanks
george



Built on fedora 23.  speexdsp-devel is required. It provides speex_echo.h .

Haven't actually run it, but it built.

Thanks for all the work. This much easier. Maybe i'll switch to 13.

sean






--
_
-- 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 a "less secure app" on google ??

2016-03-27 Thread sean darcy

On 03/27/2016 01:41 PM, Lincoln Quirk wrote:

My guess is that you should be able to use 2fa but enable this by using
an "app specific password" in your Google account (accounts.google.com
<http://accounts.google.com> > Sign-In and Security > App Passwords). I
haven't tried it with this exact setup though.

On Sun, Mar 27, 2016 at 6:13 AM, sean darcy <seandar...@gmail.com
<mailto:seandar...@gmail.com>> wrote:

To connect to google voice with xmpp, I've had to turn on the "less
secure apps" switch.

You recently changed your security settings so that your Google
Account ...@gmail.com <mailto:...@gmail.com> is no
longer protected by modern security standards.

Please be aware that it is now easier for an attacker to break
into your account.



My xmpp.conf :

type=client
serverhost=talk.google.com <http://talk.google.com>
secret=mysecret
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Not available"
timeout=5

Is there a way to configure xmpp so I don't have to turn on "less
secure apps" ?

Is this just a way of google messing with us ?

sean



But I don't use 2fa to login from a browser. It's not turned on (as far 
as I know).


sean




--
_
-- 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 a "less secure app" on google ??

2016-03-27 Thread sean darcy
To connect to google voice with xmpp, I've had to turn on the "less 
secure apps" switch.



You recently changed your security settings so that your Google Account 
...@gmail.com is no longer protected by modern security standards.

Please be aware that it is now easier for an attacker to break into your 
account.



My xmpp.conf :

type=client
serverhost=talk.google.com
secret=mysecret
priority=25
port=5222
usetls=yes
usesasl=yes
status=available
statusmessage="Not available"
timeout=5

Is there a way to configure xmpp so I don't have to turn on "less secure 
apps" ?


Is this just a way of google messing with us ?

sean


--
_
-- 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] 11.21,2 : how to transfer to Jolly Roger ?

2016-02-27 Thread sean darcy

On 02/25/2016 05:23 PM, John Kiniston wrote:

You can do this with setting up an application map using
DYNAMIC_FEATURES and enabling it on your incoming call paths.

https://wiki.asterisk.org/wiki/display/AST/Custom+Dynamic+Features

If you don't want to even answer the calls you could try doing this with
'ex-girlfriend logic', I personally have my extension set up so that
callers from outside my state go directly to Voicemail.

exten => 7132/_480NXX,1,Goto(RING)
exten => 7132/_520NXX,1,Goto(RING)
exten => 7132/_602NXX,1,Goto(RING)
exten => 7132/_623NXX,1,Goto(RING)
exten => 7132/_928NXX,1,Goto(RING)
exten => 7132,1,Voicemail(7132@{CUSTGROUP},u)
exten => 7132,n,Hangup()
exten =>
7132,n(RING),Gosub(sub-stdexten2,7132,1({CUSTGROUP},{CUSTGROUP}-operator,7133,,SIP/7132${CUSTGROUP}))

You can see I'm using a pattern-match to send calls from my in-state
area codes to the 'RING' label, other calls are answered by voicemail.


On Thu, Feb 25, 2016 at 3:13 PM, sean darcy <seandar...@gmail.com
<mailto:seandar...@gmail.com>> wrote:

I'd like to transfer all my pesky telemarketing calls to Jolly Roger .


http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html

In the middle of a call I'd hit some DTMF sequence, which would dial
Jolly Roger and transfer the call after Jolly Roger answers.

But blindtransfer requires an extension after you hear "transfer".
And I don't want the caller to hear "transfer", or hear the dialing
sequence.

Any suggestions ?

sean



Brillliant. Here's what worked.

In features.conf:

[applicationmap]
...
jollyroger = *8,peer,Dial,"MOTIF//+12146664...@voice.google.com"

In extensions.conf:

Set(_DYNAMIC_FEATURES=jollyroger)

Now eagerly awaiting a spam call !!




--
_
-- 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] 11.21,2 : how to transfer to Jolly Roger ?

2016-02-25 Thread sean darcy

I'd like to transfer all my pesky telemarketing calls to Jolly Roger .

http://www.nytimes.com/2016/02/25/fashion/a-robot-that-has-fun-at-telemarketers-expense.html

In the middle of a call I'd hit some DTMF sequence, which would dial 
Jolly Roger and transfer the call after Jolly Roger answers.


But blindtransfer requires an extension after you hear "transfer". And I 
don't want the caller to hear "transfer", or hear the dialing sequence.


Any suggestions ?

sean


--
_
-- 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] 11.21.0 : echo woes : can't install canceller (sean darcy)

2016-01-30 Thread sean darcy

On 01/29/2016 03:59 PM, Mc GRATH Ricardo wrote:

Hi Sean Darcy

Question about "the remote party always hears an echo on it's side", strange 
because eco suppression circuit is for local side.


Mc GRATH Ricardo



OK. Maybe an echo canceller won't make any difference. But why does the 
remote side _always_ hear an echo if we use a local dahdi extension, and 
_never_ when we use a local SIP extension ??


Also, and a different problem, why doesn't the echo canceller module 
load on startup ?


sean



--
_
-- 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] 11.21.0 : echo woes : can't install canceller

2016-01-28 Thread sean darcy

On 01/28/2016 03:39 PM, sean darcy wrote:

i've got calls coming into an 11.21.0 box. The internal phones are
analogue off a TDM400 board, and SIP extensions.

Using an analogue internal phone, the remote party always hears an echo
on it's side. We do not hear an echo. Doesn't matter who is the calling
party.

But if we use a SIP extension, no echo.

I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko

And requested oslec echo cancel:

grep oslec /etc/dahdi/system.conf
echocanceller=oslec,1,2,4

grep echo /etc/asterisk/chan_dahdi.conf
echocancel=yes
echocancelwhenbridged=no
echotraining=yes

but it's never loaded:

# lsmod | grep echo
[root@asterisk ~]#

dahdi_cfg -vvv
DAHDI Tools Version - 2.10.0

DAHDI Version: 2.11.0
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

I can manually insert the oslec module using modprobe.

Thats seems to work.

CLI> dahdi show version
DAHDI Version: 2.11.0 Echo Canceller: OSLEC

But it's not persistent across reboots.

sean




And even if I do manually load the oslec kernel module, I don't think 
it's actually being used


cat /proc/dahdi/1
Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)

1 WCTDM/4/0 FXOKS (In use)
2 WCTDM/4/1 FXOKS (In use)
3 WCTDM/4/2 Reserved
4 WCTDM/4/3 FXSKS (In use) RED

AFAIK, the echo canceller should show up here.

sean


--
_
-- 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] 11.21.0 : echo woes : can't install canceller

2016-01-28 Thread sean darcy
i've got calls coming into an 11.21.0 box. The internal phones are 
analogue off a TDM400 board, and SIP extensions.


Using an analogue internal phone, the remote party always hears an echo 
on it's side. We do not hear an echo. Doesn't matter who is the calling 
party.


But if we use a SIP extension, no echo.

I've built /lib/modules/4.3.3-303.fc23.x86_64/dahdi/dahdi_echocan_oslec.ko

And requested oslec echo cancel:

grep oslec /etc/dahdi/system.conf
echocanceller=oslec,1,2,4

grep echo /etc/asterisk/chan_dahdi.conf
echocancel=yes
echocancelwhenbridged=no
echotraining=yes

but it's never loaded:

# lsmod | grep echo
[root@asterisk ~]#

dahdi_cfg -vvv
DAHDI Tools Version - 2.10.0

DAHDI Version: 2.11.0
Echo Canceller(s):
Configuration
==


Channel map:


0 channels to configure.

I can manually insert the oslec module using modprobe.

Thats seems to work.

CLI> dahdi show version
DAHDI Version: 2.11.0 Echo Canceller: OSLEC

But it's not persistent across reboots.

sean


--
_
-- 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] map softphone ids to asterisk ??

2015-12-21 Thread sean darcy

In setting up the GS-Wave softphone there are two id entries:

SIP User ID
SIP Authentication ID

I would have thought SIP User ID was the devicename , i.e. [name].

Then SIP Authentication ID was defaultuser.

But not so. With

[gs_5062](cell-phones)
defaultuser=gs_62

and

SIP User ID gs-5062
SIP Authentication ID   gs-62

username mismatch, have , digest has 

So I tried:

[gs_62](cell-phones)
defaultuser=gs_5062

Registration from '' failed for 
'10.10.11.95:48115' - Username/auth name mismatch


So I commented out defaultuser:

[gs_5062](cell-phones)
;defaultuser=gs_62

No joy:

 Registration from '' failed for 
'10.10.11.95:48125' - Username/auth name mismatch



Then I took out blanked the SIP Authentication ID pn the phone.

Same result.

BUT if you set both the SIP User ID and SIP Authentication ID to the 
same value, and comment out defaultuser, it registers.



Is this what's supposed to happen ??

sean


--
_
-- 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] no ringing tone with Dial option r

2015-11-06 Thread sean darcy

On 11/04/2015 03:43 AM, Bertrand LUPART - Linkeo.com wrote:

Hello,



I'm not getting any ringing when I use option r with Dial:

Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in new 
stack


Warning, options are the 3rd arguments.

You seem to have an extra comma and a non-closed double-quote.


http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial



Here's the actual dialplan command:

exten => 
s,n(gv),Dial(motif/${MOTIF_DEFAULT}/+1${ARG1}@voice.google.com,,rTt)


No quotes. And the options are the 3rd argument, I think.

sean


--
_
-- 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] no ringing tone with Dial option r

2015-11-06 Thread sean darcy

On 11/03/2015 01:11 PM, John Kiniston wrote:

Have you checked your indications.conf? I've seen a missing or
misconfiguration in the zone definition cause this.

On Tue, Nov 3, 2015 at 11:07 AM, sean darcy <seandar...@gmail.com
<mailto:seandar...@gmail.com>> wrote:

On 11/01/2015 12:38 PM, sean darcy wrote:

I'm not getting any ringing when I use option r with Dial:

Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com
<http://voice.google.com>,,rTt") in
new stack

Otherwise all works. The call goes through, good audio.

sean


FWIW, 11.18.0 on Fedora 22.


sean



AFAIK, I've never touched indications.conf . Not even sure what zone 
definitions are in indications.


Also, now on 11.20.0. Same problem.

sean



--
_
-- 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] no ringing tone with Dial option r

2015-11-06 Thread sean darcy

On 11/04/2015 03:40 AM, A J Stiles wrote:

On Tuesday 03 Nov 2015, sean darcy wrote:

On 11/01/2015 12:38 PM, sean darcy wrote:

I'm not getting any ringing when I use option r with Dial:

Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in
new stack

Otherwise all works. The call goes through, good audio.

sean


FWIW, 11.18.0 on Fedora 22.

sean


Make sure you have an Answer(), or some command that does an implicit
Answer(), somewhere in the dialplan before the Dial() statement with the r
option.  Been bitten that way before .



Me too. I put it as the first command.

-- Starting simple switch on 'DAHDI/1-1'
-- Executing [@internal:1] Answer("DAHDI/1-1", "") in new stack
..
-- Executing [s@DialOut:15] Dial("DAHDI/1-1", 
"motif/8447/+1@voice.google.com,,Ttr") in new stack


Maybe I need to put another Answer() in the DialOut context. Would 2 
Answer() cause a problem?


sean


--
_
-- 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] no ringing tone with Dial option r

2015-11-03 Thread sean darcy

On 11/01/2015 12:38 PM, sean darcy wrote:

I'm not getting any ringing when I use option r with Dial:

Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in
new stack

Otherwise all works. The call goes through, good audio.

sean



FWIW, 11.18.0 on Fedora 22.

sean



--
_
-- 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] no ringing tone with Dial option r

2015-11-01 Thread sean darcy

I'm not getting any ringing when I use option r with Dial:

Dial("DAHDI/1-1", "motif/8447/+1@voice.google.com,,rTt") in 
new stack


Otherwise all works. The call goes through, good audio.

sean


--
_
-- 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] howto copy a voicemail message to another machine ?

2015-06-16 Thread sean darcy
My asterisk server is in the cloud. Figuring out how to send an email is 
too much brain damage. So i can't use the email feature that's built 
into voicemail.


What I want to do is execute a remote command with the voicemail as an 
argument.  The remote machine command would email the message.


I'm thinking of:

same =n,VoiceMail(vm,u)
same =n,System(ssh myserver emailVM  '_THE_VOICEMAIL_MESSAGE_')

What variables can I use for _THE_VOICEMAIL_MESSAGE_

Or is this better done with externcmd in voicemail.conf ?:

externcmd = ssh myserver emailVM  '_THE_VOICEMAIL_MESSAGE_'

But same question. And can externcmd take arguments ?

sean



--
_
-- 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] howto copy a voicemail message to another machine ?

2015-06-16 Thread sean darcy

On 06/16/2015 11:52 AM, D'Arcy J.M. Cain wrote:

On Tue, 16 Jun 2015 11:35:26 -0400
sean darcy seandar...@gmail.com wrote:

My asterisk server is in the cloud. Figuring out how to send an email
is too much brain damage. So i can't use the email feature that's
built into voicemail.


Really?  That was one of the first things I did when I learned
Asterisk.  It was dead simple.  Rather than creating some sort of Rube
Goldberg machine to send email why don't you explain what you tried and
where you ran into problems?

In case it helps, here are my general settings in voicemail.conf:

[general]
attach=yes
maxsilence=10
serveremail=n...@vex.net
format=wav49
fromstring=Vex.Net Voice Mail
nextaftercmd=yes
forcename=yes
pollmailboxes=yes
pollfreq=5

And for each extension I have this (sanitized):

1000 = 1234,D'Arcy,da...@example.com

That's the extension, PIN, name and email address.

There's no problem setting up vm on *. I can't use email off the 
instance, since the assigned ip address doesn't have a PTR.


It looks too much like spam. The mail relays drop it.

sean


--
_
-- 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] more vm woes

2015-06-16 Thread sean darcy

on 11.17.1

Trying to debug vm. as suggested by voip-info:

mailcmd=cat  /tmp/astvm-mail

and the mailbox is:

1143 = ,sean,s...@company.com,,|tz=eastern|attach=yes|saycid=yes


But there [s nothing in /tmp. The messages are in 
/var/spool/asterisk/voicemail/43-vm/1143/ .


Any help appreciated.

sean


--
_
-- 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] Free Fax for Asterisk : registered, but license not effective

2015-06-06 Thread sean darcy

On 06/05/2015 08:15 PM, sean darcy wrote:

I've been having problems segfaulting when receiving faxes. So I tried
FFA. I used the register utility and my license is in /var/lib/asterisk :
ls /var/lib/asterisk/licenses/ -l
total 4
-rw-r--r--. 1 root root 325 Jun  5 22:41 FFA-DNCXXX.lic

but:

Executing [17772822954@FaxIncoming:4]
ReceiveFAX(SIP/ec2faxcall18-0002,
/var/spool/asterisk/fax/20150605_2352.tif,df) in new stack
 -- Channel 'SIP/ec2faxcall18-0002' receiving FAX
'/var/spool/asterisk/fax/20150605_2352.tif'
[Jun  5 23:52:39] WARNING[12361][C-0002]: res_fax_digium.c:1604
dgm_fax_reserve: Cannot reserve FAX session - session limit exceeded
(max: 0).
[Jun  5 23:52:39] ERROR[12361][C-0002]: res_fax.c:1906
receivefax_exec: Unable to reserve FAX session.

and

fax show licenses
Fax Licensing Information
==
Free fax licenses: 0
Total licensed ports: 0

??

sean



I'm on 11.17.1, and installed: res_fax_digium-11.0_1.3.1-core2_64.tar.gz

jay


--
_
-- 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] 11.17.1 : ReceiveFax then signal 11 ??

2015-06-05 Thread sean darcy

dialplan

[FaxIncoming]

exten=s,1,NoOp(Incoming fax on 46-va)

same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
 same=n,Answer()
 same=n,ReceiveFAX(${FAXFILE}.tif,df)
 same=n,Hangup()
 exten=h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} 
FAXPAGES: ${FAXPAGES} @ bitrate ${FAXBITRATE})
  same=n,System(scp ${FAXFILE}.tif 
asterisk@asterisk:/home/asterisk/ec2fax/ )
  same=n,System(ssh asterisk@asterisk 
/home/asterisk/bin/email-ec2fax.sh  ${FAXFILE} ${CALLERID(name)} 
${CALLERID(num)  )


cli
.
-- Executing [s@FaxIncoming:3] Answer(SIP/ec2faxcall17-, 
) in new stack
-- Executing [s@FaxIncoming:4] 
ReceiveFAX(SIP/ec2faxcall17-, 
/var/spool/asterisk/fax/20150605_2137.tif,df) in new stack
-- Channel 'SIP/ec2faxcall17-' receiving FAX 
'/var/spool/asterisk/fax/20150605_2137.tif'

ip-172-31-53-29*CLI
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups

syslog

asterisk[11540]: -- Executing [s@FaxIncoming:3] 
Answer(SIP/ec2faxcall17-, ) in new stack
asterisk[11540]: -- Executing [s@FaxIncoming:4] 
ReceiveFAX(SIP/ec2faxcall17-, /var/spool/a...ew stack
asterisk[11540]: -- Channel 'SIP/ec2faxcall17-' receiving 
FAX '/var/spool/asterisk/fax/20150605_2137.tif'
systemd[1]: asterisk.service: main process exited, code=killed, 
status=11/SEGV
asterisk[11587]: Unable to connect to remote asterisk (does 
/run/asterisk/asterisk.ctl exist?


Doesn't Signal 11 mean accessing prohibited ram ?

Does it matter this on an ec2 instance?

Thanks for any help.

sean




--
_
-- 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] Free Fax for Asterisk : registered, but license not effective

2015-06-05 Thread sean darcy
I've been having problems segfaulting when receiving faxes. So I tried 
FFA. I used the register utility and my license is in /var/lib/asterisk :

ls /var/lib/asterisk/licenses/ -l
total 4
-rw-r--r--. 1 root root 325 Jun  5 22:41 FFA-DNCXXX.lic

but:

Executing [17772822954@FaxIncoming:4] 
ReceiveFAX(SIP/ec2faxcall18-0002, 
/var/spool/asterisk/fax/20150605_2352.tif,df) in new stack
-- Channel 'SIP/ec2faxcall18-0002' receiving FAX 
'/var/spool/asterisk/fax/20150605_2352.tif'
[Jun  5 23:52:39] WARNING[12361][C-0002]: res_fax_digium.c:1604 
dgm_fax_reserve: Cannot reserve FAX session - session limit exceeded 
(max: 0).
[Jun  5 23:52:39] ERROR[12361][C-0002]: res_fax.c:1906 
receivefax_exec: Unable to reserve FAX session.


and

fax show licenses
Fax Licensing Information
==
Free fax licenses: 0
Total licensed ports: 0

??

sean


--
_
-- 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] did i miss the memo on asterisk devel ?

2015-06-02 Thread sean darcy

I usually lurk on the asterisk devel list to see what's going on.

No posts for a week or two. Has the list moved ?

sean


--
_
-- 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] dahdi 2.10.1 build fails

2015-06-01 Thread sean darcy

On fedora 21, trying to build dahdi for kernel 4.0.4.

gcc-4.9.2-6.fc21.x86_64

make -C /lib/modules/4.0.4-202.fc21.x86_64/build 
SUBDIRS=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi 
DAHDI_INCLUDE=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/include 
DAHDI_MODULES_EXTRA=  HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m

make[1]: Entering directory '/usr/src/kernels/4.0.4-202.fc21.x86_64'
  CC [M] 
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.o
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c: 
In function 'set_spanno_and_basechan':
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7056:2: 
error: void value not ignored as it ought to be

  dahdi_dev_dbg(ASSIGN, span_device(span),
  ^
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c: 
In function '_assign_spanno_and_basechan':
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7080:2: 
error: void value not ignored as it ought to be

  dahdi_dev_dbg(ASSIGN, span_device(span),
  ^
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7100:2: 
error: void value not ignored as it ought to be

  dahdi_dev_dbg(ASSIGN, span_device(span),
  ^
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c: 
In function '_check_spanno_and_basechan':
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7150:2: 
error: void value not ignored as it ought to be

  dahdi_dev_dbg(ASSIGN, span_device(span),
  ^
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7156:3: 
error: void value not ignored as it ought to be

   dahdi_dev_dbg(ASSIGN, span_device(span),
   ^
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7184:2: 
error: void value not ignored as it ought to be

  dahdi_dev_dbg(ASSIGN, span_device(span),
  ^


--
_
-- 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] dahdi 2.10.1 build fails

2015-06-01 Thread sean darcy

On 06/01/2015 04:49 PM, Shaun Ruffell wrote:

On Mon, Jun 01, 2015 at 04:24:31PM -0400, sean darcy wrote:

On fedora 21, trying to build dahdi for kernel 4.0.4.

gcc-4.9.2-6.fc21.x86_64

make -C /lib/modules/4.0.4-202.fc21.x86_64/build 
SUBDIRS=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi 
DAHDI_INCLUDE=/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/include
DAHDI_MODULES_EXTRA=  HOTPLUG_FIRMWARE=yes modules
DAHDI_BUILD_ALL=m
make[1]: Entering directory '/usr/src/kernels/4.0.4-202.fc21.x86_64'
   CC [M] 
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.o
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:
In function 'set_spanno_and_basechan':
/home/asterisk/rpmbuild/BUILD/dahdi-linux-2.10.1/drivers/dahdi/dahdi-base.c:7056:2:
error: void value not ignored as it ought to be
   dahdi_dev_dbg(ASSIGN, span_device(span),

...

I've not tried building against Fedora Core 21 myself, but I believe
this is already fixed on DAHDI's master branch [1].

Russ Meyerriecks has posted a comment on DAHLIN-346 [2] with a link
for how to install from source while waiting for the next release of
dahdi-linux.

[1] 
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=1cc0ad510acd404e63923ed3062b93
[2] https://issues.asterisk.org/jira/browse/DAHLIN-346


Yup. that worked.

Thanks.

Maybe time for the next release ??

sean


--
_
-- 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] timeout on non-critical invite spamming log

2015-04-26 Thread sean darcy

On 11.17.1:

The cli and the log are full of these warnings:

 WARNING[12110]: chan_sip.c:4086 retrans_pkt: Timeout on 849421411 on 
non-critical invite transaction.


The number is a random 9-10 digits.

What causes them? How do I stop them ?

sean


--
_
-- 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] which libsrtp ?

2015-03-03 Thread sean darcy
I've been having some issues with srtp. so I checked which version of 
libsrtp I built asterisk 11.6 against. I'm on fedora 21, so 
libsrtp-1.4.4-13.20101004cvs.fc21.x86_64.


From https://github.com/cisco/libsrtp it seems that latest release is 
1.5.1, released a couple of weeks ago.


I'm not a fan of the bleeding edge, but using a version 4+ years old 
seems strange even to me. But, on the other hand, it seems to Work For Me.


Anybody using 1.5.1 ?

sean


--
_
-- 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] Upgrade to Fedora 21, now gv requires rtp ?

2015-03-01 Thread sean darcy
I just upgraded to fedora 21. I'm running asterisk 11.6.0. All works 
with Fedora 20.



-- Executing [s@DialOut:15] Dial(DAHDI/1-1, 
motif/8447/+1212xxxy...@voice.google.com,,rTt) in new stack
[Mar  1 21:24:06] ERROR[2477][C-]: rtp_engine.c:259 
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
[Mar  1 21:24:06] ERROR[2477][C-]: chan_motif.c:1820 
jingle_request: Unable to create Jingle session on endpoint '8447'




any help appreciated.

sean


--
_
-- 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] 11.5.0: blindxfer problems

2014-12-24 Thread sean darcy

On 12/22/2014 05:38 PM, Richard Mudgett wrote:



On Mon, Dec 22, 2014 at 4:00 PM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:

snip

How do I enable DTMF logging?

logger set level DEBUG
No such command 'logger set level DEBUG' (type 'core show help
logger set level' for other possible commands)

didn't work, even though:

help logger
logger mute Toggle logging output to a console
  logger reload Reopens the log files
  logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging
level for this console


The help syntax string is truncated because of the number
of options and the column length.  The command is:
logger set level DTMF

The full help syntax string is:
logger set level {DEBUG|NOTICE|WARNING|ERROR|VERBOSE|DTMF} {on|off}

Richard


Thanks.

cli logger shows _no_ dtmf events, yet recognizes *2  !!

logger set level dtmf on
Logger status for 'DTMF' has been set to 'on'.

 here I first hit *7, then hit *2 

-- Started music on hold, class 'default', on 
Motif/+12036258...@voice.google.com-8bf9

-- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
.

sean




--
_
-- 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] 11.5.0: blindxfer problems

2014-12-22 Thread sean darcy

On 12/21/2014 11:09 AM, sean darcy wrote:

On 12/21/2014 04:42 AM, Patrick Beaumont wrote:

Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.




On 20/12/2014 20:52, sean darcy seandar...@gmail.com wrote:


On 12/20/2014 03:22 PM, sean darcy wrote:

On 12/19/2014 09:42 AM, Rusty Newton wrote:

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy seandar...@gmail.com
wrote:

I've got a confbridge set up which works if dialed locally:

 -- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new
stack
  -- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new
stack
  -- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in
new stack
  -- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...


extensions.conf:

[globals]
...
GOTO_ON_BLINDXFR=internal,266,1

features.conf:

[featuremap]
blindxfer = #1

But:

-- Executing [s@DialOut:14] Dial(DAHDI/1-1,
motif//+12345678...@voice.google.com,,rTt) in new stack
  -- Called motif//+12345678...@voice.google.com
  -- Motif/+12345678...@voice.google.com-688c is proceeding
passing it to
DAHDI/1-1
  -- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
  -- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
  -- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
  -- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')

I'm expecting the blind transfer to GOTO internal,266,1.

If I input 266 at the transfer dial tone, the blind transfer occurs.

Do I have this set up incorrectly?



https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Var

iables


${GOTO_ON_BLINDXFR} - Transfer to the specified
context/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)

Try using ^ characters as it mentions there.



Thanks for the response, but no joy:


   == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'

DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550
builtin_blindtransfer: No digits dialed.


sean




I also tried setting up a transfer as an applicationmap.

conference = *7,peer/both,ConfBridge,1

Seems to load:

features reload
   == Parsing '/etc/asterisk/features.conf': Found
   == Registered Feature 'conference'
   == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7'

but when the caller dials *7, there's no action, Nothing in the cli. The
dtmf is just sent to the callee.

Also tried having the callee dial *7, same result.

Any help appreciated.



OK. I'll figure out DTMF logging, but notice asterisk does recognize
both #1 (blindxfer) and *2 (atxfer), so it recognizes DTMF tones.

sean



How do I enable DTMF logging?

logger set level DEBUG
No such command 'logger set level DEBUG' (type 'core show help logger 
set level' for other possible commands)


didn't work, even though:

help logger
   logger mute Toggle logging output to a console
 logger reload Reopens the log files
 logger rotate Rotates and reopens the log files
logger set level {DEBUG|NOTICE Enables/Disables a specific logging level 
for this console



I tried core set debug 10 , but that captured no DTMF.

sean






--
_
-- 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] Smartphone Mobility App?

2014-12-22 Thread sean darcy

On 12/19/2014 09:29 AM, chris wrote:

Anyone found any good smartphone apps that connect with their asterisk
boxes that provides basic mobility features?

The main problem we are trying to solve is when our staff  forward to
their cell phones they cant distinguish if the call was directed at
their cell phone or the business DID.



To solve this, can you mess with the callerid ? For a forwarded call:

Set(CALLERID(name)=DID_${CALLERID(name)})

sean


We also would like to give user ability to control DND and forwarding of
their extension from the smartphone.

I know there are many cloud service providers with a offering like this
but we are not looking to change our service infrastructure but rather
looking for just a software product that connects to our existing
asterisk systems and provides this functionality.

We would ideally like something for both iphone and android but the
immediate need is for iPhone

Curious to hear what people have tried, their experiences, etc.

We are open to both free/open source as well as commercial software as
long as it is multitenant or scalable beyond single server.

TIA,
chris






--
_
-- 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] 11.5.0: blindxfer problems

2014-12-21 Thread sean darcy

On 12/21/2014 04:42 AM, Patrick Beaumont wrote:

Have you enabled DTMF logging and seen the DTMF codes being recognised by
Asterisk? I had a bunch of soft phones that I had to change to using “sip
info” for the DTMF signalling as the RFC signalling was not always being
recognised. This would cause transfers to appear as if the user had not
dialled any digits.




On 20/12/2014 20:52, sean darcy seandar...@gmail.com wrote:


On 12/20/2014 03:22 PM, sean darcy wrote:

On 12/19/2014 09:42 AM, Rusty Newton wrote:

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy seandar...@gmail.com
wrote:

I've got a confbridge set up which works if dialed locally:

 -- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new stack
  -- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new
stack
  -- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in
new stack
  -- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...


extensions.conf:

[globals]
...
GOTO_ON_BLINDXFR=internal,266,1

features.conf:

[featuremap]
blindxfer = #1

But:

-- Executing [s@DialOut:14] Dial(DAHDI/1-1,
motif//+12345678...@voice.google.com,,rTt) in new stack
  -- Called motif//+12345678...@voice.google.com
  -- Motif/+12345678...@voice.google.com-688c is proceeding
passing it to
DAHDI/1-1
  -- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
  -- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
  -- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
  -- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')

I'm expecting the blind transfer to GOTO internal,266,1.

If I input 266 at the transfer dial tone, the blind transfer occurs.

Do I have this set up incorrectly?



https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Var
iables


${GOTO_ON_BLINDXFR} - Transfer to the specified
context/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)

Try using ^ characters as it mentions there.



Thanks for the response, but no joy:


   == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'

DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550
builtin_blindtransfer: No digits dialed.


sean




I also tried setting up a transfer as an applicationmap.

conference = *7,peer/both,ConfBridge,1

Seems to load:

features reload
   == Parsing '/etc/asterisk/features.conf': Found
   == Registered Feature 'conference'
   == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7'

but when the caller dials *7, there's no action, Nothing in the cli. The
dtmf is just sent to the callee.

Also tried having the callee dial *7, same result.

Any help appreciated.



OK. I'll figure out DTMF logging, but notice asterisk does recognize 
both #1 (blindxfer) and *2 (atxfer), so it recognizes DTMF tones.


sean



--
_
-- 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] 11.5.0: blindxfer problems

2014-12-20 Thread sean darcy

On 12/19/2014 09:42 AM, Rusty Newton wrote:

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy seandar...@gmail.com wrote:

I've got a confbridge set up which works if dialed locally:

-- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new stack
 -- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new stack
 -- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in new stack
 -- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...


extensions.conf:

[globals]
...
GOTO_ON_BLINDXFR=internal,266,1

features.conf:

[featuremap]
blindxfer = #1

But:

-- Executing [s@DialOut:14] Dial(DAHDI/1-1,
motif//+12345678...@voice.google.com,,rTt) in new stack
 -- Called motif//+12345678...@voice.google.com
 -- Motif/+12345678...@voice.google.com-688c is proceeding passing it to
DAHDI/1-1
 -- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
 -- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
 -- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
 -- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')

I'm expecting the blind transfer to GOTO internal,266,1.

If I input 266 at the transfer dial tone, the blind transfer occurs.

Do I have this set up incorrectly?


https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables

${GOTO_ON_BLINDXFR} - Transfer to the specified
context/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)

Try using ^ characters as it mentions there.



Thanks for the response, but no joy:


 == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'

DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550 
builtin_blindtransfer: No digits dialed.



sean


--
_
-- 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] 11.5.0: blindxfer problems

2014-12-20 Thread sean darcy

On 12/20/2014 03:22 PM, sean darcy wrote:

On 12/19/2014 09:42 AM, Rusty Newton wrote:

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy seandar...@gmail.com wrote:

I've got a confbridge set up which works if dialed locally:

-- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new stack
 -- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new
stack
 -- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in
new stack
 -- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...


extensions.conf:

[globals]
...
GOTO_ON_BLINDXFR=internal,266,1

features.conf:

[featuremap]
blindxfer = #1

But:

-- Executing [s@DialOut:14] Dial(DAHDI/1-1,
motif//+12345678...@voice.google.com,,rTt) in new stack
 -- Called motif//+12345678...@voice.google.com
 -- Motif/+12345678...@voice.google.com-688c is proceeding
passing it to
DAHDI/1-1
 -- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
 -- Started music on hold, class 'default', on
Motif/+1234567...@voice.google.com-688c
 -- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550
builtin_blindtransfer: No digits dialed.
 -- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')

I'm expecting the blind transfer to GOTO internal,266,1.

If I input 266 at the transfer dial tone, the blind transfer occurs.

Do I have this set up incorrectly?


https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables


${GOTO_ON_BLINDXFR} - Transfer to the specified
context/extension/priority after a blind transfer (use ^ characters in
place of | to separate context/extension/priority when setting this
variable from the dialplan)

Try using ^ characters as it mentions there.



Thanks for the response, but no joy:


  == Setting global variable 'GOTO_ON_BLINDXFER' to 'internal^266^1'

DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 20 15:12:03] WARNING[12336][C-0012]: features.c:2550
builtin_blindtransfer: No digits dialed.


sean




I also tried setting up a transfer as an applicationmap.

conference = *7,peer/both,ConfBridge,1

Seems to load:

features reload
  == Parsing '/etc/asterisk/features.conf': Found
  == Registered Feature 'conference'
  == Mapping Feature 'conference' to app 'ConfBridge(1)' with code '*7'

but when the caller dials *7, there's no action, Nothing in the cli. The 
dtmf is just sent to the callee.


Also tried having the callee dial *7, same result.

Any help appreciated.


--
_
-- 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] 11.5.0: blindxfer problems

2014-12-17 Thread sean darcy

I've got a confbridge set up which works if dialed locally:

   -- Executing [266@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [266@internal:2] SendDTMF(DAHDI/1-1, 1) in new stack
-- Executing [266@internal:3] ConfBridge(DAHDI/1-1, 1) in new stack
-- DAHDI/1-1 Playing 'conf-onlyperson.ulaw' (language 'en')
...


extensions.conf:

[globals]
...
GOTO_ON_BLINDXFR=internal,266,1

features.conf:

[featuremap]
blindxfer = #1

But:

-- Executing [s@DialOut:14] Dial(DAHDI/1-1, 
motif//+12345678...@voice.google.com,,rTt) in new stack

-- Called motif//+12345678...@voice.google.com
-- Motif/+12345678...@voice.google.com-688c is proceeding passing 
it to DAHDI/1-1

-- Motif/+12345678...@voice.google.com-688c answered DAHDI/1-1
-- Started music on hold, class 'default', on 
Motif/+1234567...@voice.google.com-688c

-- DAHDI/1-1 Playing 'pbx-transfer.ulaw' (language 'en')
[Dec 17 09:46:59] WARNING[19083][C-00be]: features.c:2550 
builtin_blindtransfer: No digits dialed.

-- DAHDI/1-1 Playing 'pbx-invalid.ulaw' (language 'en')

I'm expecting the blind transfer to GOTO internal,266,1.

If I input 266 at the transfer dial tone, the blind transfer occurs.

Do I have this set up incorrectly?

sean


--
_
-- 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] On Fedora, kernel update resets /var/run/asterisk owner to root.root

2014-12-02 Thread sean darcy
On Fedora 20, every time the kernel updates, /var/run/asterisk owner is 
set to root.root.  I'm running asterisk under user asterisk.


Is there any way to keep /var/run/asterisk as asterisk.asterisk. Or do I 
find a new place to put asterisk.pid?


sean


--
_
-- 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] On Fedora, kernel update resets /var/run/asterisk owner to root.root

2014-12-02 Thread sean darcy

On 12/02/2014 02:46 PM, Jeffrey Ollie wrote:

On Tue, Dec 2, 2014 at 1:22 PM, sean darcy seandar...@gmail.com wrote:


Or do I
find a new place to put asterisk.pid?


Also, if you use the native systemd unit file, you no longer need a
PID file, although you still need /run/asterisk to store the control
socket.



So systemd is taking over the galaxy.

Put asterisk.conf in /etc/tmpfiles.d, and all worked. It needs to be 
included in the rpm.


I still need ctl so I can access * remotely.

Thanks for the help.

sean


--
_
-- 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 : SILK codec ?

2014-10-29 Thread sean darcy

Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?

sean


--
_
-- 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 13 : SILK codec ?

2014-10-29 Thread sean darcy

On 10/29/2014 08:06 PM, Matthew Jordan wrote:

On Wed, Oct 29, 2014 at 5:16 PM, sean darcy seandar...@gmail.com wrote:

Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13?



codec_silk for Asterisk 12 will most likely not work in Asterisk 13. A
number of performance improvements in the media handling in Asterisk
required some codec compatibility changes.

I would expect said modules to be available in the next few weeks.


Great. Thanks.
sean


--
_
-- 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] 11.13.1: unable to load sip.conf (or iax )

2014-10-26 Thread sean darcy

On 10/24/2014 03:24 PM, Jeffrey Ollie wrote:

On Fri, Oct 24, 2014 at 1:47 PM, sean darcy seandar...@gmail.com wrote:

On 10/24/2014 02:21 PM, Jeffrey Ollie wrote:


restorecon -rv /etc/asterisk


I'd never have guessed.


Yeah, if you mv the data instead of cp the data from one place to
the other, the SElinux labels don't get updated.  I like SElinux, but
it would be nice if there were better error messages...

Although, if you're on Fedora 20, this is a pretty good description of
what was going wrong and how to diagnose/solve it:

http://danwalsh.livejournal.com/65777.html

Thanks for the reference, but in my case systemctl status asterisk, 
which I did try, had no reference to selinux.


sean


--
_
-- 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] 11.13.1: unable to load sip.conf (or iax )

2014-10-24 Thread sean darcy

On 10/23/2014 01:19 PM, sean darcy wrote:

On 10/23/2014 11:26 AM, sean darcy wrote:

Running 11.13.1 on Fedora.

This is a new install, but a copy of a previous - working -install.

module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to
load config sip.conf

I don't think it's permissions:

ls -ld /etc/asterisk /etc/asterisk/sip*
drwxr-x---. 4 asterisk asterisk  4096 Oct 23 00:34 /etc/asterisk
-rw-r-. 1 asterisk asterisk  3588 Oct 22 18:37 /etc/asterisk/sip.conf
-rw-r-. 1 asterisk asterisk 91033 Oct 23 00:28
/etc/asterisk/sip.conf.rpmnew
-rw-r-. 1 asterisk asterisk   790 Oct 23 00:28
/etc/asterisk/sip_notify.conf

ps aux | grep asterisk
asterisk   294  0.1  5.5 1076736 33364 ?   Ssl  14:36   0:03
/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf

The sip module itself is loaded:

module show like chan_sip
Module Description Use Count
chan_sip.soSession Initiation Protocol (SIP) 0
1 modules loaded

I've tried my old config, and just the sip.conf.sample. Same result.

FWIW:

  ls -l /usr/lib64/asterisk/modules/chan*
-rwxr-xr-x. 1 root root  72808 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_agent.so
-rwxr-xr-x. 1 root root  16032 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_bridge.so
-rwxr-xr-x. 1 root root 347920 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_iax2.so
-rwxr-xr-x. 1 root root  41888 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_local.so
-rwxr-xr-x. 1 root root 118144 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_mgcp.so
-rwxr-xr-x. 1 root root  67424 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_motif.so
-rwxr-xr-x. 1 root root  11936 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_multicast_rtp.so
-rwxr-xr-x. 1 root root  44392 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_phone.so
-rwxr-xr-x. 1 root root 755296 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_sip.so


Any help appreciated.


sean




Weirdness:

made iax.conf.simple:

[general]
autokill=yes

[idefisk]
type=friend
host=dynamic
context=phones

(extra credit for remembering the source)

 module unload chan_iax2.so
Unable to unload resource chan_iax2.so
Command 'module unload chan_iax2.so' failed.
[Oct 23 16:53:26] WARNING[669]: loader.c:571 ast_unload_resource: Unload
failed, 'chan_iax2.so' is not loaded.
  module load chan_iax2.so
Unable to load module chan_iax2.so
Command 'module load chan_iax2.so' failed.
[Oct 23 16:53:36] ERROR[669]: chan_iax2.c:13488 set_config: Unable to
load config iax.conf

But then:

cp -a iax.conf.simple iax.conf
cp: overwrite ‘iax.conf’? y
  ls -l iax*
-rw-r-. 1 asterisk asterisk   74 Oct 23 16:52 iax.conf
-rw-r-. 1 asterisk asterisk  652 Oct 22 18:37 iax.conf.real
-rw-r-. 1 asterisk asterisk   74 Oct 23 16:52 iax.conf.simple

  module load chan_iax2.so
Loaded chan_iax2.so

cp iax.conf.real iax.conf
cp: overwrite ‘iax.conf’? y

module unload chan_iax2.so
Unloaded chan_iax2.so
 module load chan_iax2.so
Loaded chan_iax2.so

So the simple config will load.  Then if I unload it, and the real
config will load !!

This approach also works for sip.conf, but now have another problem : it
won't recognize any of the #includes. For instance:

module load chan_sip.so
Unable to load module chan_sip.so
Command 'module load chan_sip.so' failed.
SIP channel loading...
[Oct 23 17:13:43] ERROR[669]: config.c:1549 process_text_line: The file
'/etc/asterisk/exts/droid.sip.conf' was listed as a #include but it does
not exist.
[Oct 23 17:13:43] ERROR[669]: chan_sip.c:31461 reload_config: Contents
of sip.conf are invalid and cannot be parsed

grep exts/droid.sip  sip.conf
#include /etc/asterisk/exts/droid.sip.conf

ls -l /etc/asterisk/exts/droid.sip.conf
-rw-r--r--. 1 asterisk asterisk 316 Oct 22 18:37
/etc/asterisk/exts/droid.sip.conf

I also tried relative addressing,  exts/droid.sip.conf , same problem.

And, of course, all this works on the 11.10.2 server.

sean



Weirder yet:

ls -ld /etc/asterisk/test /etc/asterisk/exts
drwxr-xr-x. 3 644 asterisk 4096 Oct 24 16:41 /etc/asterisk/exts
drwxr-xr-x. 2 644 asterisk 4096 Oct 24 16:44 /etc/asterisk/test

cp exts/droid.sip.conf test/droid2.sip.conf

ls -l /etc/asterisk/exts/droid.sip.conf /etc/asterisk/test/droid2.sip.conf
-rw-r--r--. 1 644 asterisk 316 Oct 22 18:37 
/etc/asterisk/exts/droid.sip.conf
-rw-r--r--. 1 644 asterisk 316 Oct 24 16:44 
/etc/asterisk/test/droid2.sip.conf


grep droid  sip.conf
#include test/droid2.sip.conf
#include exts/droid.sip.conf

module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
  == Parsing '/etc/asterisk/sip.conf': Found
  == Parsing '/etc/asterisk/test/droid2.sip.conf': Found
[Oct 24 16:47:39] ERROR[2743]: config.c:1549 process_text_line: The file 
'exts/droid.sip.conf' was listed as a #include but it does not exist

Re: [asterisk-users] 11.13.1: unable to load sip.conf (or iax )

2014-10-24 Thread sean darcy

On 10/24/2014 02:21 PM, Jeffrey Ollie wrote:

restorecon -rv /etc/asterisk

I'd never have guessed.

Thanks. I owe you a beer. At least one.

sean


--
_
-- 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] 11.13.1: unable to load sip.conf (or iax )

2014-10-23 Thread sean darcy

Running 11.13.1 on Fedora.

This is a new install, but a copy of a previous - working -install.

module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to 
load config sip.conf


I don't think it's permissions:

ls -ld /etc/asterisk /etc/asterisk/sip*
drwxr-x---. 4 asterisk asterisk  4096 Oct 23 00:34 /etc/asterisk
-rw-r-. 1 asterisk asterisk  3588 Oct 22 18:37 /etc/asterisk/sip.conf
-rw-r-. 1 asterisk asterisk 91033 Oct 23 00:28 
/etc/asterisk/sip.conf.rpmnew
-rw-r-. 1 asterisk asterisk   790 Oct 23 00:28 
/etc/asterisk/sip_notify.conf


ps aux | grep asterisk
asterisk   294  0.1  5.5 1076736 33364 ?   Ssl  14:36   0:03 
/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf


The sip module itself is loaded:

module show like chan_sip
Module Description 
Use Count
chan_sip.soSession Initiation Protocol (SIP) 
0

1 modules loaded

I've tried my old config, and just the sip.conf.sample. Same result.

FWIW:

 ls -l /usr/lib64/asterisk/modules/chan*
-rwxr-xr-x. 1 root root  72808 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_agent.so
-rwxr-xr-x. 1 root root  16032 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_bridge.so
-rwxr-xr-x. 1 root root 347920 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_iax2.so
-rwxr-xr-x. 1 root root  41888 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_local.so
-rwxr-xr-x. 1 root root 118144 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_mgcp.so
-rwxr-xr-x. 1 root root  67424 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_motif.so
-rwxr-xr-x. 1 root root  11936 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_multicast_rtp.so
-rwxr-xr-x. 1 root root  44392 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_phone.so
-rwxr-xr-x. 1 root root 755296 Oct 23 00:29 
/usr/lib64/asterisk/modules/chan_sip.so



Any help appreciated.


sean


--
_
-- 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] 11.13.1: unable to load sip.conf (or iax )

2014-10-23 Thread sean darcy

On 10/23/2014 11:26 AM, sean darcy wrote:

Running 11.13.1 on Fedora.

This is a new install, but a copy of a previous - working -install.

module load chan_sip
Unable to load module chan_sip
Command 'module load chan_sip' failed.
SIP channel loading...
[Oct 23 14:46:08] NOTICE[669]: chan_sip.c:31438 reload_config: Unable to
load config sip.conf

I don't think it's permissions:

ls -ld /etc/asterisk /etc/asterisk/sip*
drwxr-x---. 4 asterisk asterisk  4096 Oct 23 00:34 /etc/asterisk
-rw-r-. 1 asterisk asterisk  3588 Oct 22 18:37 /etc/asterisk/sip.conf
-rw-r-. 1 asterisk asterisk 91033 Oct 23 00:28
/etc/asterisk/sip.conf.rpmnew
-rw-r-. 1 asterisk asterisk   790 Oct 23 00:28
/etc/asterisk/sip_notify.conf

ps aux | grep asterisk
asterisk   294  0.1  5.5 1076736 33364 ?   Ssl  14:36   0:03
/usr/sbin/asterisk -f -C /etc/asterisk/asterisk.conf

The sip module itself is loaded:

module show like chan_sip
Module Description Use Count
chan_sip.soSession Initiation Protocol (SIP) 0
1 modules loaded

I've tried my old config, and just the sip.conf.sample. Same result.

FWIW:

  ls -l /usr/lib64/asterisk/modules/chan*
-rwxr-xr-x. 1 root root  72808 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_agent.so
-rwxr-xr-x. 1 root root  16032 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_bridge.so
-rwxr-xr-x. 1 root root 347920 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_iax2.so
-rwxr-xr-x. 1 root root  41888 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_local.so
-rwxr-xr-x. 1 root root 118144 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_mgcp.so
-rwxr-xr-x. 1 root root  67424 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_motif.so
-rwxr-xr-x. 1 root root  11936 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_multicast_rtp.so
-rwxr-xr-x. 1 root root  44392 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_phone.so
-rwxr-xr-x. 1 root root 755296 Oct 23 00:29
/usr/lib64/asterisk/modules/chan_sip.so


Any help appreciated.


sean




Weirdness:

made iax.conf.simple:

[general]
autokill=yes

[idefisk]
type=friend
host=dynamic
context=phones

(extra credit for remembering the source)

module unload chan_iax2.so
Unable to unload resource chan_iax2.so
Command 'module unload chan_iax2.so' failed.
[Oct 23 16:53:26] WARNING[669]: loader.c:571 ast_unload_resource: Unload 
failed, 'chan_iax2.so' is not loaded.

 module load chan_iax2.so
Unable to load module chan_iax2.so
Command 'module load chan_iax2.so' failed.
[Oct 23 16:53:36] ERROR[669]: chan_iax2.c:13488 set_config: Unable to 
load config iax.conf


But then:

cp -a iax.conf.simple iax.conf
cp: overwrite ‘iax.conf’? y
 ls -l iax*
-rw-r-. 1 asterisk asterisk   74 Oct 23 16:52 iax.conf
-rw-r-. 1 asterisk asterisk  652 Oct 22 18:37 iax.conf.real
-rw-r-. 1 asterisk asterisk   74 Oct 23 16:52 iax.conf.simple

 module load chan_iax2.so
Loaded chan_iax2.so

cp iax.conf.real iax.conf
cp: overwrite ‘iax.conf’? y

module unload chan_iax2.so
Unloaded chan_iax2.so
module load chan_iax2.so
Loaded chan_iax2.so

So the simple config will load.  Then if I unload it, and the real 
config will load !!


This approach also works for sip.conf, but now have another problem : it 
won't recognize any of the #includes. For instance:


module load chan_sip.so
Unable to load module chan_sip.so
Command 'module load chan_sip.so' failed.
SIP channel loading...
[Oct 23 17:13:43] ERROR[669]: config.c:1549 process_text_line: The file 
'/etc/asterisk/exts/droid.sip.conf' was listed as a #include but it does 
not exist.
[Oct 23 17:13:43] ERROR[669]: chan_sip.c:31461 reload_config: Contents 
of sip.conf are invalid and cannot be parsed


grep exts/droid.sip  sip.conf
#include /etc/asterisk/exts/droid.sip.conf

ls -l /etc/asterisk/exts/droid.sip.conf
-rw-r--r--. 1 asterisk asterisk 316 Oct 22 18:37 
/etc/asterisk/exts/droid.sip.conf


I also tried relative addressing,  exts/droid.sip.conf , same problem.

And, of course, all this works on the 11.10.2 server.

sean



--
_
-- 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] On kernel 3.16.2 : dahdi_rec: Invalid argument

2014-09-13 Thread sean darcy
On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against 
it. dahdi show channels works fine, but when I try to place a call:


chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument

Any help appreciated.

sean


--
_
-- 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


  1   2   3   4   5   >