Re: [asterisk-dev] Asterisk now available with bundled pjproject!

2016-03-19 Thread George Joseph 
On Wed, Mar 16, 2016 at 9:30 AM, Ross Beer  wrote:

>
> Package matching libsrtp-devel-1.4.4-10.20101004cvs.el7.x86_64 already
> installed.
>
> pjproject builds correct with the following:
>
> ./configure CFLAGS="-DNDEBUG" --prefix=/usr --libdir=/usr/lib64
> --enable-epoll --enable-shared --disable-video --disable-sound
> --disable-opencore-amr
>
>

​You're compiling with pjproject's internal libsrtp implementation.  Try
with --with-external-srtp and see what happens.
libsrtp-devel in Fedora is already at 1.5.4 so maybe it's a version thing.
  There was a ticket open with pjproject for this exact problem but it was
implemented 2 years ago.  Maybe it's not quite right.  I'll check.



>
>
> --
> From: george.jos...@fairview5.com
> Date: Wed, 16 Mar 2016 09:09:30 -0600
>
> To: asterisk-dev@lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!
>
>
>
> On Wed, Mar 16, 2016 at 8:47 AM, Ross Beer  wrote:
>
> After running install_prereq I get the following error:
>
>
> [GENERATE] libasteriskpj.exports
>[LD] libasteriskpj.o -> libasteriskpj.so.2
>[LN] libasteriskpj.so.2 -> libasteriskpj.so
>[LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o
> ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o
> astobj2_hash.o astobj2_rbtree.o audiohook.o autochan.o autoservice.o
> backtrace.o bridge.o bridge_after.o bridge_basic.o bridge_channel.o
> bridge_roles.o bucket.o callerid.o ccss.o cdr.o cel.o channel.o
> channel_internal_api.o chanvars.o cli.o codec.o codec_builtin.o config.o
> config_options.o core_local.o core_unreal.o crypt.o data.o datastore.o db.o
> devicestate.o dial.o dns.o dnsmgr.o dsp.o endpoints.o enum.o event.o
> features.o features_config.o file.o fixedjitterbuf.o format.o
> format_cache.o format_cap.o format_compatibility.o frame.o framehook.o
> fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o
> indications.o io.o jitterbuf.o json.o loader.o lock.o logger.o manager.o
> manager_bridges.o manager_channels.o manager_endpoints.o manager_mwi.o
> manager_system.o max_forwards.o md5.o media_index.o message.o mixmonitor.o
> named_acl.o netsock.o netsock2.o optional_api.o parking.o pbx.o pbx_app.o
> pbx_builtins.o pbx_functions.o pbx_hangup_handler.o pbx_switch.o
> pbx_timing.o pbx_variables.o pickup.o plc.o poll.o presencestate.o
> privacy.o rtp_engine.o say.o sched.o sdp_srtp.o security_events.o sem.o
> sha1.o sip_api.o slinfactory.o smoother.o sorcery.o sounds_index.o srv.o
> stasis.o stasis_bridges.o stasis_cache.o stasis_cache_pattern.o
> stasis_channels.o stasis_endpoints.o stasis_message.o
> stasis_message_router.o stasis_system.o stdtime/localtime.o strcompat.o
> strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o
> threadpool.o threadstorage.o timing.o translate.o udptl.o ulaw.o uri.o
> utils.o uuid.o version.o xml.o xmldoc.o   -> asterisk
> ./libasteriskpj.so: undefined reference to `srtp_deinit'
> collect2: error: ld returned 1 exit status
> make[1]: *** [asterisk] Error 1
> make: *** [main] Error 2
>
>
>
> ​Ok, that's weird.  I take it you have libsrtp-devel installed
> (install_prereq should have done it)?  What version?
> Can you build ​pjproject from source normally?  When you do, do you use
> --with-external-srtp?
>
>
>
>
>
>
>
> --
> From: george.jos...@fairview5.com
> Date: Wed, 16 Mar 2016 07:37:44 -0600
> To: asterisk-dev@lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!
>
>
>
>
> On Wed, Mar 16, 2016 at 5:41 AM, Ross Beer  wrote:
>
> Hi,
>
> I just attempted to install with the bundled pjproject however the
> following error stopped the build:
>
> Generating embedded module rules ...
>[CC] astdb2sqlite3.c -> astdb2sqlite3.o
>[LD] astdb2sqlite3.o db1-ast/libdb1.a -> astdb2sqlite3
>[CC] astdb2bdb.c -> astdb2bdb.o
>[LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
> [pjproject]  Making dependencies
> [pjproject]  Compiling libs
> [pjproject]  Generating symbols
> [pjproject]  Compiling apps
> [pjproject]  Compiling python bindings
> make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
> make[1]: *** [pjproject] Error 2
> make: *** [third-party] Error 2
>
>
> ​I'll bet you don't have the python development libraries installed.  The
> install_prereq script was updated to include python-devel or python-dev
> depending on the distribution.​
>
>
> Kind regards,
>
> ROss
>
> --
> From: george.jos...@fairview5.com
> Date: Mon, 7 Mar 2016 12:28:23 -0700
> To: asterisk-dev@lists.digium.com; asterisk-us...@lists.digium.com
> Subject: [asterisk-dev] Asterisk now available with bundled pjproject!
>
>
> The current Asterisk 13 and master git branches have a new feature that
> will be included in 13.8.0:  The ability to compile and run Asterisk with a
> bundled version of

Re: [asterisk-dev] Asterisk now available with bundled pjproject!

2016-03-19 Thread Ross Beer 
Hi,
 
I just attempted to install with the bundled pjproject however the following 
error stopped the build:
 
Generating embedded module rules ...
   [CC] astdb2sqlite3.c -> astdb2sqlite3.o
   [LD] astdb2sqlite3.o db1-ast/libdb1.a -> astdb2sqlite3
   [CC] astdb2bdb.c -> astdb2bdb.o
   [LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
[pjproject]  Making dependencies
[pjproject]  Compiling libs
[pjproject]  Generating symbols
[pjproject]  Compiling apps
[pjproject]  Compiling python bindings
make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
make[1]: *** [pjproject] Error 2
make: *** [third-party] Error 2

Kind regards,
 
ROss
 
From: george.jos...@fairview5.com
Date: Mon, 7 Mar 2016 12:28:23 -0700
To: asterisk-dev@lists.digium.com; asterisk-us...@lists.digium.com
Subject: [asterisk-dev] Asterisk now available with bundled pjproject!

The current Asterisk 13 and master git branches have a new feature that will be 
included in 13.8.0:  The ability to compile and run Asterisk with a bundled 
version of pjproject.
​​
Why would you want to do this?  Several reasons:
Predictability:  When built with the ​bundled pjproject, you're always certain 
of the version you're running against, no matter where it's installed.
Scalability:  The default pjproject configuration is optimized for client 
applications. The bundled version's configuration is optimized for server use.
Usability:  Several feature patches, which have been submitted upstream to 
pjproject but not yet released, have been included in the bundled version.
Safety:  If a security or critical issue is identified in pjproject, it can be 
patched and made available with a new release of Asterisk instead of ​having to 
​waiting for a new release of pjproject​​.Maintainability:  You don't need to 
build and install separate packages.
Supportability:  When asking others for help, there's no question about which 
version of pjproject you're using and what options it was compiled with.
Compatibility:  This is especially important from a development perspective 
because it means we can be sure that new pjproject APIs that have been 
introduced​,​ or old ones that have been deprecated​,​ are handled and tested 
appropriately in Asterisk.
Reliability:  You can be sure that Asterisk was tested against the bundled 
version.

So now that you're sold, here's how you use it:

All you have to do is add the "--with-pjproject-bundled" option to your 
./configure command line and remove any other "--with-pjproject" option you may 
have specified.  The configure and make processes will download the correct 
version of pjproject, patch it, configure it, build it and finally link 
Asterisk to it statically.  No changes in runtime configuration are required.

Still not sold?  The default behavior hasn't changed so as long as you haven't 
specified "--with-pjproject-bundled", your build and deploy process remains as 
is.

PLEASE TRY THIS!!  I'd love some feedback BEFORE 13.8.0 is released.


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[asterisk-dev] Add .gitignore (libss7[master])

2016-03-19 Thread Richard Mudgett 
Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/2421

Change subject: Add .gitignore
..

Add .gitignore

Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6
---
A .gitignore
1 file changed, 11 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/libss7 refs/changes/21/2421/1

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 000..ad4dc18
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,11 @@
+*.o
+*.o.d
+*.lo
+*.so
+*.a
+libss7.so.*
+parser_debug
+ss7linktest
+ss7test
+version.c
+

-- 
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Gerrit-Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6
Gerrit-PatchSet: 1
Gerrit-Project: libss7
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett 

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[asterisk-dev] Add .gitignore (libpri[1.4])

2016-03-19 Thread Richard Mudgett 
Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/2423

Change subject: Add .gitignore
..

Add .gitignore

Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62
---
A .gitignore
1 file changed, 12 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/libpri refs/changes/23/2423/1

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 000..78b897b
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,12 @@
+*.o
+*.o.d
+*.lo
+*.so
+*.a
+libpri.so.*
+pridump
+pritest
+rosetest
+testprilib
+version.c
+

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Gerrit-MessageType: newchange
Gerrit-Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62
Gerrit-PatchSet: 1
Gerrit-Project: libpri
Gerrit-Branch: 1.4
Gerrit-Owner: Richard Mudgett 

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Re: [asterisk-dev] Asterisk now available with bundled pjproject!

2016-03-19 Thread George Joseph 
On Wed, Mar 16, 2016 at 10:25 AM, George Joseph  wrote:

>
>
> On Wed, Mar 16, 2016 at 9:30 AM, Ross Beer  wrote:
>
>>
>> Package matching libsrtp-devel-1.4.4-10.20101004cvs.el7.x86_64 already
>> installed.
>>
>> pjproject builds correct with the following:
>>
>> ./configure CFLAGS="-DNDEBUG" --prefix=/usr --libdir=/usr/lib64
>> --enable-epoll --enable-shared --disable-video --disable-sound
>> --disable-opencore-amr
>>
>>
>
> ​You're compiling with pjproject's internal libsrtp implementation.  Try
> with --with-external-srtp and see what happens.
> libsrtp-devel in Fedora is already at 1.5.4 so maybe it's a version thing.
>   There was a ticket open with pjproject for this exact problem but it was
> implemented 2 years ago.  Maybe it's not quite right.  I'll check.
>
>
​I just tested on my CentOS7 VM with the same version of libsrtp and didn't
have any problems.  Maybe try a distclean and reconfigure?



>
>
>>
>>
>> --
>> From: george.jos...@fairview5.com
>> Date: Wed, 16 Mar 2016 09:09:30 -0600
>>
>> To: asterisk-dev@lists.digium.com
>> Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!
>>
>>
>>
>> On Wed, Mar 16, 2016 at 8:47 AM, Ross Beer  wrote:
>>
>> After running install_prereq I get the following error:
>>
>>
>> [GENERATE] libasteriskpj.exports
>>[LD] libasteriskpj.o -> libasteriskpj.so.2
>>[LN] libasteriskpj.so.2 -> libasteriskpj.so
>>[LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o
>> ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o
>> astobj2_hash.o astobj2_rbtree.o audiohook.o autochan.o autoservice.o
>> backtrace.o bridge.o bridge_after.o bridge_basic.o bridge_channel.o
>> bridge_roles.o bucket.o callerid.o ccss.o cdr.o cel.o channel.o
>> channel_internal_api.o chanvars.o cli.o codec.o codec_builtin.o config.o
>> config_options.o core_local.o core_unreal.o crypt.o data.o datastore.o db.o
>> devicestate.o dial.o dns.o dnsmgr.o dsp.o endpoints.o enum.o event.o
>> features.o features_config.o file.o fixedjitterbuf.o format.o
>> format_cache.o format_cap.o format_compatibility.o frame.o framehook.o
>> fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o
>> indications.o io.o jitterbuf.o json.o loader.o lock.o logger.o manager.o
>> manager_bridges.o manager_channels.o manager_endpoints.o manager_mwi.o
>> manager_system.o max_forwards.o md5.o media_index.o message.o mixmonitor.o
>> named_acl.o netsock.o netsock2.o optional_api.o parking.o pbx.o pbx_app.o
>> pbx_builtins.o pbx_functions.o pbx_hangup_handler.o pbx_switch.o
>> pbx_timing.o pbx_variables.o pickup.o plc.o poll.o presencestate.o
>> privacy.o rtp_engine.o say.o sched.o sdp_srtp.o security_events.o sem.o
>> sha1.o sip_api.o slinfactory.o smoother.o sorcery.o sounds_index.o srv.o
>> stasis.o stasis_bridges.o stasis_cache.o stasis_cache_pattern.o
>> stasis_channels.o stasis_endpoints.o stasis_message.o
>> stasis_message_router.o stasis_system.o stdtime/localtime.o strcompat.o
>> strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o
>> threadpool.o threadstorage.o timing.o translate.o udptl.o ulaw.o uri.o
>> utils.o uuid.o version.o xml.o xmldoc.o   -> asterisk
>> ./libasteriskpj.so: undefined reference to `srtp_deinit'
>> collect2: error: ld returned 1 exit status
>> make[1]: *** [asterisk] Error 1
>> make: *** [main] Error 2
>>
>>
>>
>> ​Ok, that's weird.  I take it you have libsrtp-devel installed
>> (install_prereq should have done it)?  What version?
>> Can you build ​pjproject from source normally?  When you do, do you use
>> --with-external-srtp?
>>
>>
>>
>>
>>
>>
>>
>> --
>> From: george.jos...@fairview5.com
>> Date: Wed, 16 Mar 2016 07:37:44 -0600
>> To: asterisk-dev@lists.digium.com
>> Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!
>>
>>
>>
>>
>> On Wed, Mar 16, 2016 at 5:41 AM, Ross Beer  wrote:
>>
>> Hi,
>>
>> I just attempted to install with the bundled pjproject however the
>> following error stopped the build:
>>
>> Generating embedded module rules ...
>>[CC] astdb2sqlite3.c -> astdb2sqlite3.o
>>[LD] astdb2sqlite3.o db1-ast/libdb1.a -> astdb2sqlite3
>>[CC] astdb2bdb.c -> astdb2bdb.o
>>[LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
>> [pjproject]  Making dependencies
>> [pjproject]  Compiling libs
>> [pjproject]  Generating symbols
>> [pjproject]  Compiling apps
>> [pjproject]  Compiling python bindings
>> make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
>> make[1]: *** [pjproject] Error 2
>> make: *** [third-party] Error 2
>>
>>
>> ​I'll bet you don't have the python development libraries installed.  The
>> install_prereq script was updated to include python-devel or python-dev
>> depending on the distribution.​
>>
>>
>> Kind regards,
>>
>> ROss
>>
>> --
>> From: george.jos...@fairview5.com

[asterisk-dev] q931.c: Substitute PROGRESS for DISCONNECT with progress ind... (libpri[1.4])

2016-03-19 Thread Richard Mudgett 
Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/2425

Change subject: q931.c: Substitute PROGRESS for DISCONNECT with progress 
indicator #8
..

q931.c: Substitute PROGRESS for DISCONNECT with progress indicator #8

When the pri_set_inbanddisconnect() option is enabled and the call has not
been answered when a DISCONNECT with progress indicator #8 (Inband audio
present) is received, then report the event as a PROGRESS with progress
indicator #8 (Inband audio present) instead.  Substituting a PROGRESS
event allows the upper layer to open the media path if it isn't already
open so the user can hear the inband audio message.

PRI-180
Reported by: Alexandr Dranchuk

Change-Id: I62313bf9cc1d2f3b0231f0c07a784717ddba0415
---
M q931.c
1 file changed, 34 insertions(+), 1 deletion(-)


  git pull ssh://gerrit.asterisk.org:29418/libpri refs/changes/25/2425/1

diff --git a/q931.c b/q931.c
index 0e7ff79..69c210e 100644
--- a/q931.c
+++ b/q931.c
@@ -8726,6 +8726,7 @@
int res;
int changed;
int mand_cause;
+   enum Q931_CALL_STATE ourcallstate_orig;
struct apdu_event *cur = NULL;
struct pri_subcommand *subcmd;
struct q931_call *master_call;
@@ -9265,14 +9266,46 @@
}
}
 
+   ourcallstate_orig = c->ourcallstate;
UPDATE_OURCALLSTATE(ctrl, c, 
Q931_CALL_STATE_DISCONNECT_INDICATION);
c->peercallstate = Q931_CALL_STATE_DISCONNECT_REQUEST;
c->sendhangupack = 1;
 
/* wait for a RELEASE so that sufficient time has passed
   for the inband audio to be heard */
-   if (ctrl->acceptinbanddisconnect && (c->progressmask & 
PRI_PROG_INBAND_AVAILABLE))
+   if (ctrl->acceptinbanddisconnect
+   && (c->progressmask & PRI_PROG_INBAND_AVAILABLE)) {
+   switch (ourcallstate_orig) {
+   case Q931_CALL_STATE_CALL_INITIATED:
+   case Q931_CALL_STATE_OVERLAP_SENDING:
+   case Q931_CALL_STATE_OUTGOING_CALL_PROCEEDING:
+   case Q931_CALL_STATE_CALL_DELIVERED:
+   /*
+* Open the media path if it isn't already open 
so
+* the user can hear the inband audio.
+*/
+   if (ctrl->debug & PRI_DEBUG_Q931_STATE) {
+   pri_message(ctrl, "Report the 
DISCONNECT as a PROGRESS instead.\n");
+   }
+   ctrl->ev.e = PRI_EVENT_PROGRESS;
+   ctrl->ev.proceeding.cause = c->cause;
+   ctrl->ev.proceeding.subcmds = >subcmds;
+   ctrl->ev.proceeding.channel = 
q931_encode_channel(c);
+   ctrl->ev.proceeding.progress = c->progress;
+   ctrl->ev.proceeding.progressmask = 
c->progressmask;
+   ctrl->ev.proceeding.cref = c->cr;
+   ctrl->ev.proceeding.call = c->master_call;
+   return Q931_RES_HAVEEVENT;
+   default:
+   break;
+   }
+   /*
+* Suppress reporting DISCONNECT to the upper layer.  
The
+* media path should already be open and we cannot 
report
+* a PROGRESS at this time anyway.
+*/
break;
+   }
 
/* Return such an event */
ctrl->ev.e = PRI_EVENT_HANGUP_REQ;

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Gerrit-Project: libpri
Gerrit-Branch: 1.4
Gerrit-Owner: Richard Mudgett 

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Re: [asterisk-dev] Issues with ARI python example for recording

2016-03-19 Thread Nitesh Bansal 
Hi,

It may well be an issue with Asterisk.
Here is the HTTP traffic dump


{"state":"queued","format":"wav","name":"voicemail/
6/1458310870.38","target_uri":"channel:1458310870.16"}POST
/ari/recordings/live/voicemail/%206/1458310870.38/stop HTTP/1.1
Host: 37.139.25.109:8088
Content-Length: 0
Authorization: Basic YXN0ZXJpc2s6YXN0ZXJpc2s=
Accept-Encoding: gzip, deflate, compress
Accept: */*
User-Agent: python-requests/2.2.1 CPython/2.7.6 Linux/3.13.0-77-generic

HTTP/1.1 404 Not Found
Server: Asterisk/13.7.2
Date: Fri, 18 Mar 2016 14:21:13 GMT
Cache-Control: no-cache, no-store
Content-type: application/json
Content-Length: 32

{"message":"Resource not found"}



The weird thing is that recording is saved in Asterisk nicely, no issues
with saving the recordings.
I tried changing the code example and modified the recording path in
example to be extension dialed (6) and it worked, no HTTP error
response.
In the failure case, recording path is 'voicemail/%206/1458310870.38'.

Thanks,
Nitesh




On Fri, Mar 18, 2016 at 2:57 PM, Mark Michelson 
wrote:

> On 03/18/2016 06:18 AM, Nitesh Bansal wrote:
>
>> Hello,
>>
>> I'm using the latest version of ari-py library.
>> I'm trying the following demo
>> https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+1+-+Recording
>> .
>>
>> Everything is setup correctly, I can call Asterisk, but when I press the
>> DTMF '#' to
>> stop the recording, my python library throws an exception.
>>
>>
>> Entering recording state
>> Recording voicemail at voicemail/ 6/1458299017.29
>> stopping recording LiveRecording(voicemail/ 6/1458299017.29)
>> ERROR:ari.client:Event listener threw exception
>> Traceback (most recent call last):
>>   File "build/bdist.linux-x86_64/egg/ari/client.py", line 100, in __run
>> callback(msg_json, *args, **kwargs)
>>   File "build/bdist.linux-x86_64/egg/ari/client.py", line 198, in
>> extract_objects
>> event_cb(obj, event, *args, **kwargs)
>>   File "build/bdist.linux-x86_64/egg/ari/model.py", line 181, in fn_filter
>> fn(objects, event, *args, **kwargs)
>>   File
>> "/root/asterisk_ari/ari-py/examples/ari_bridges/recording_demo/recording_state.py",
>> line 39, in on_dtmf
>> self.recording.stop()
>>   File "build/bdist.linux-x86_64/egg/ari/model.py", line 155, in
>> enrich_operation
>> return promote(self.client, oper(**kwargs), oper.json)
>>   File "build/bdist.linux-x86_64/egg/ari/model.py", line 354, in promote
>> resp.raise_for_status()
>>   File "/usr/lib/python2.7/dist-packages/requests/models.py", line 773,
>> in raise_for_status
>> raise HTTPError(http_error_msg, response=self)
>> HTTPError: 404 Client Error: Not Found
>>
>> It is throwing error on the line
>> 'self.recording.stop'
>>
>> Personally, I can't see anything wrong with this code, any ideas what I'm
>> doing wrong or
>> is there any bug in the ARI lib?
>>
>> Thanks,
>> Nitesh
>>
>> Hi.
>
> It's hard to tell exactly what's going wrong here, but Asterisk is
> responding to the HTTP request with a 404. This likely means that Asterisk
> can't find the recording in question. Although it is also possible that the
> URI being passed by python to Asterisk does not "resolve" as expected. If
> you look at the HTTP traffic, you may be able to tell more clearly why the
> 404 is being sent.
>
> It's certainly possible there's a bug in Asterisk, but I would be more
> willing to bet that there's some sort of error in the python example.
> Although, like you, I don't immediately see the problem in the python code.
>
> Mark Michelson
>
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[asterisk-dev] Add .gitignore (libpri[master])

2016-03-19 Thread Richard Mudgett 
Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/2424

Change subject: Add .gitignore
..

Add .gitignore

Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62
---
A .gitignore
1 file changed, 12 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/libpri refs/changes/24/2424/1

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 000..78b897b
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,12 @@
+*.o
+*.o.d
+*.lo
+*.so
+*.a
+libpri.so.*
+pridump
+pritest
+rosetest
+testprilib
+version.c
+

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Gerrit-MessageType: newchange
Gerrit-Change-Id: I11ac3b47a9d5d0a0c1ea4559280b75ef5d866d62
Gerrit-PatchSet: 1
Gerrit-Project: libpri
Gerrit-Branch: master
Gerrit-Owner: Richard Mudgett 

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[asterisk-dev] Add .gitignore (libss7[2.0])

2016-03-19 Thread Richard Mudgett 
Richard Mudgett has uploaded a new change for review.

  https://gerrit.asterisk.org/2420

Change subject: Add .gitignore
..

Add .gitignore

Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6
---
A .gitignore
1 file changed, 11 insertions(+), 0 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/libss7 refs/changes/20/2420/1

diff --git a/.gitignore b/.gitignore
new file mode 100644
index 000..ad4dc18
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,11 @@
+*.o
+*.o.d
+*.lo
+*.so
+*.a
+libss7.so.*
+parser_debug
+ss7linktest
+ss7test
+version.c
+

-- 
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I068e59b57102551e3ef7c10eb5774ac071a7c5b6
Gerrit-PatchSet: 1
Gerrit-Project: libss7
Gerrit-Branch: 2.0
Gerrit-Owner: Richard Mudgett 

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Re: [asterisk-dev] Asterisk now available with bundled pjproject!

2016-03-19 Thread George Joseph 
On Wed, Mar 16, 2016 at 8:47 AM, Ross Beer  wrote:

> After running install_prereq I get the following error:
>
>
> [GENERATE] libasteriskpj.exports
>[LD] libasteriskpj.o -> libasteriskpj.so.2
>[LN] libasteriskpj.so.2 -> libasteriskpj.so
>[LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o
> ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o
> astobj2_hash.o astobj2_rbtree.o audiohook.o autochan.o autoservice.o
> backtrace.o bridge.o bridge_after.o bridge_basic.o bridge_channel.o
> bridge_roles.o bucket.o callerid.o ccss.o cdr.o cel.o channel.o
> channel_internal_api.o chanvars.o cli.o codec.o codec_builtin.o config.o
> config_options.o core_local.o core_unreal.o crypt.o data.o datastore.o db.o
> devicestate.o dial.o dns.o dnsmgr.o dsp.o endpoints.o enum.o event.o
> features.o features_config.o file.o fixedjitterbuf.o format.o
> format_cache.o format_cap.o format_compatibility.o frame.o framehook.o
> fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o
> indications.o io.o jitterbuf.o json.o loader.o lock.o logger.o manager.o
> manager_bridges.o manager_channels.o manager_endpoints.o manager_mwi.o
> manager_system.o max_forwards.o md5.o media_index.o message.o mixmonitor.o
> named_acl.o netsock.o netsock2.o optional_api.o parking.o pbx.o pbx_app.o
> pbx_builtins.o pbx_functions.o pbx_hangup_handler.o pbx_switch.o
> pbx_timing.o pbx_variables.o pickup.o plc.o poll.o presencestate.o
> privacy.o rtp_engine.o say.o sched.o sdp_srtp.o security_events.o sem.o
> sha1.o sip_api.o slinfactory.o smoother.o sorcery.o sounds_index.o srv.o
> stasis.o stasis_bridges.o stasis_cache.o stasis_cache_pattern.o
> stasis_channels.o stasis_endpoints.o stasis_message.o
> stasis_message_router.o stasis_system.o stdtime/localtime.o strcompat.o
> strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o
> threadpool.o threadstorage.o timing.o translate.o udptl.o ulaw.o uri.o
> utils.o uuid.o version.o xml.o xmldoc.o   -> asterisk
> ./libasteriskpj.so: undefined reference to `srtp_deinit'
> collect2: error: ld returned 1 exit status
> make[1]: *** [asterisk] Error 1
> make: *** [main] Error 2
>
>
>
​Ok, that's weird.  I take it you have libsrtp-devel installed
(install_prereq should have done it)?  What version?
Can you build ​pjproject from source normally?  When you do, do you use
--with-external-srtp?






>
> --
> From: george.jos...@fairview5.com
> Date: Wed, 16 Mar 2016 07:37:44 -0600
> To: asterisk-dev@lists.digium.com
> Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!
>
>
>
>
> On Wed, Mar 16, 2016 at 5:41 AM, Ross Beer  wrote:
>
> Hi,
>
> I just attempted to install with the bundled pjproject however the
> following error stopped the build:
>
> Generating embedded module rules ...
>[CC] astdb2sqlite3.c -> astdb2sqlite3.o
>[LD] astdb2sqlite3.o db1-ast/libdb1.a -> astdb2sqlite3
>[CC] astdb2bdb.c -> astdb2bdb.o
>[LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
> [pjproject]  Making dependencies
> [pjproject]  Compiling libs
> [pjproject]  Generating symbols
> [pjproject]  Compiling apps
> [pjproject]  Compiling python bindings
> make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
> make[1]: *** [pjproject] Error 2
> make: *** [third-party] Error 2
>
>
> ​I'll bet you don't have the python development libraries installed.  The
> install_prereq script was updated to include python-devel or python-dev
> depending on the distribution.​
>
>
> Kind regards,
>
> ROss
>
> --
> From: george.jos...@fairview5.com
> Date: Mon, 7 Mar 2016 12:28:23 -0700
> To: asterisk-dev@lists.digium.com; asterisk-us...@lists.digium.com
> Subject: [asterisk-dev] Asterisk now available with bundled pjproject!
>
>
> The current Asterisk 13 and master git branches have a new feature that
> will be included in 13.8.0:  The ability to compile and run Asterisk with a
> bundled version of pjproject.
> ​​
>
> Why would you want to do this?  Several reasons:
>
>- Predictability:  When built with the
>​bundled
> pjproject, you're always certain of the version you're running
>against, no matter where it's installed.
>- Scalability:  The default pjproject configuration is optimized for
>client applications. The bundled version's configuration is optimized for
>server use.
>- Usability:  Several feature patches, which have been submitted
>upstream to pjproject but not yet released, have been included in the
>bundled version.
>- Safety:  If a security or critical issue is identified in pjproject,
>it can be patched and made available with a new release of Asterisk instead
>of
>​having to ​
>waiting for a new release of pjproject
>​​
>.
>- Maintainability:  You don't need to build and install separate
>packages.
>- Supportability:  When asking others for help, there's no

Re: [asterisk-dev] Issues with ARI python example for recording

2016-03-19 Thread Mark Michelson 

On 03/18/2016 06:18 AM, Nitesh Bansal wrote:

Hello,

I'm using the latest version of ari-py library.
I'm trying the following demo 
https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+1+-+Recording.


Everything is setup correctly, I can call Asterisk, but when I press 
the DTMF '#' to

stop the recording, my python library throws an exception.


Entering recording state
Recording voicemail at voicemail/ 6/1458299017.29
stopping recording LiveRecording(voicemail/ 6/1458299017.29)
ERROR:ari.client:Event listener threw exception
Traceback (most recent call last):
  File "build/bdist.linux-x86_64/egg/ari/client.py", line 100, in __run
callback(msg_json, *args, **kwargs)
  File "build/bdist.linux-x86_64/egg/ari/client.py", line 198, in 
extract_objects

event_cb(obj, event, *args, **kwargs)
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 181, in fn_filter
fn(objects, event, *args, **kwargs)
  File 
"/root/asterisk_ari/ari-py/examples/ari_bridges/recording_demo/recording_state.py", 
line 39, in on_dtmf

self.recording.stop()
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 155, in 
enrich_operation

return promote(self.client, oper(**kwargs), oper.json)
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 354, in promote
resp.raise_for_status()
  File "/usr/lib/python2.7/dist-packages/requests/models.py", line 
773, in raise_for_status

raise HTTPError(http_error_msg, response=self)
HTTPError: 404 Client Error: Not Found

It is throwing error on the line
'self.recording.stop'

Personally, I can't see anything wrong with this code, any ideas what 
I'm doing wrong or

is there any bug in the ARI lib?

Thanks,
Nitesh


Hi.

It's hard to tell exactly what's going wrong here, but Asterisk is 
responding to the HTTP request with a 404. This likely means that 
Asterisk can't find the recording in question. Although it is also 
possible that the URI being passed by python to Asterisk does not 
"resolve" as expected. If you look at the HTTP traffic, you may be able 
to tell more clearly why the 404 is being sent.


It's certainly possible there's a bug in Asterisk, but I would be more 
willing to bet that there's some sort of error in the python example. 
Although, like you, I don't immediately see the problem in the python code.


Mark Michelson

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Re: [asterisk-dev] Asterisk now available with bundled pjproject!

2016-03-19 Thread Ross Beer 
Hi,
 
This is now working as expected, however the build failed as the 'bzip2' and 
'patch' package needed to be installed. Could this be added to the 
'install_prereq' script to negate the issue?
 
Regards,
 
Ross
 
From: george.jos...@fairview5.com
Date: Wed, 16 Mar 2016 14:27:10 -0600
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!



On Wed, Mar 16, 2016 at 10:25 AM, George Joseph  
wrote:


On Wed, Mar 16, 2016 at 9:30 AM, Ross Beer  wrote:



 
Package matching libsrtp-devel-1.4.4-10.20101004cvs.el7.x86_64 already 
installed.
 
pjproject builds correct with the following:
 
./configure CFLAGS="-DNDEBUG" --prefix=/usr --libdir=/usr/lib64 --enable-epoll 
--enable-shared --disable-video --disable-sound --disable-opencore-amr
 

​You're compiling with pjproject's internal libsrtp implementation.  Try with 
--with-external-srtp and see what happens.libsrtp-devel in Fedora is already at 
1.5.4 so maybe it's a version thing.   There was a ticket open with pjproject 
for this exact problem but it was implemented 2 years ago.  Maybe it's not 
quite right.  I'll check.

​I just tested on my CentOS7 VM with the same version of libsrtp and didn't 
have any problems.  Maybe try a distclean and reconfigure?
  
 
From: george.jos...@fairview5.com
Date: Wed, 16 Mar 2016 09:09:30 -0600
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!



On Wed, Mar 16, 2016 at 8:47 AM, Ross Beer  wrote:



After running install_prereq I get the following error:
 
[GENERATE] libasteriskpj.exports
   [LD] libasteriskpj.o -> libasteriskpj.so.2
   [LN] libasteriskpj.so.2 -> libasteriskpj.so
   [LD] abstract_jb.o acl.o adsi.o alaw.o aoc.o app.o ast_expr2.o ast_expr2f.o 
asterisk.o astfd.o astmm.o astobj2.o astobj2_container.o astobj2_hash.o 
astobj2_rbtree.o audiohook.o autochan.o autoservice.o backtrace.o bridge.o 
bridge_after.o bridge_basic.o bridge_channel.o bridge_roles.o bucket.o 
callerid.o ccss.o cdr.o cel.o channel.o channel_internal_api.o chanvars.o cli.o 
codec.o codec_builtin.o config.o config_options.o core_local.o core_unreal.o 
crypt.o data.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o 
endpoints.o enum.o event.o features.o features_config.o file.o fixedjitterbuf.o 
format.o format_cache.o format_cap.o format_compatibility.o frame.o framehook.o 
fskmodem.o global_datastores.o hashtab.o heap.o http.o image.o indications.o 
io.o jitterbuf.o json.o loader.o lock.o logger.o manager.o manager_bridges.o 
manager_channels.o manager_endpoints.o manager_mwi.o manager_system.o 
max_forwards.o md5.o media_index.o message.o mixmonitor.o named_acl.o netsock.o 
netsock2.o optional_api.o parking.o pbx.o pbx_app.o pbx_builtins.o 
pbx_functions.o pbx_hangup_handler.o pbx_switch.o pbx_timing.o pbx_variables.o 
pickup.o plc.o poll.o presencestate.o privacy.o rtp_engine.o say.o sched.o 
sdp_srtp.o security_events.o sem.o sha1.o sip_api.o slinfactory.o smoother.o 
sorcery.o sounds_index.o srv.o stasis.o stasis_bridges.o stasis_cache.o 
stasis_cache_pattern.o stasis_channels.o stasis_endpoints.o stasis_message.o 
stasis_message_router.o stasis_system.o stdtime/localtime.o strcompat.o 
strings.o stun.o syslog.o taskprocessor.o tcptls.o tdd.o term.o test.o 
threadpool.o threadstorage.o timing.o translate.o udptl.o ulaw.o uri.o utils.o 
uuid.o version.o xml.o xmldoc.o   -> asterisk
./libasteriskpj.so: undefined reference to `srtp_deinit'
collect2: error: ld returned 1 exit status
make[1]: *** [asterisk] Error 1
make: *** [main] Error 2


​Ok, that's weird.  I take it you have libsrtp-devel installed (install_prereq 
should have done it)?  What version?Can you build ​pjproject from source 
normally?  When you do, do you use --with-external-srtp?



  
From: george.jos...@fairview5.com
Date: Wed, 16 Mar 2016 07:37:44 -0600
To: asterisk-dev@lists.digium.com
Subject: Re: [asterisk-dev] Asterisk now available with bundled pjproject!



On Wed, Mar 16, 2016 at 5:41 AM, Ross Beer  wrote:



Hi,
 
I just attempted to install with the bundled pjproject however the following 
error stopped the build:
 
Generating embedded module rules ...
   [CC] astdb2sqlite3.c -> astdb2sqlite3.o
   [LD] astdb2sqlite3.o db1-ast/libdb1.a -> astdb2sqlite3
   [CC] astdb2bdb.c -> astdb2bdb.o
   [LD] astdb2bdb.o db1-ast/libdb1.a -> astdb2bdb
[pjproject]  Making dependencies
[pjproject]  Compiling libs
[pjproject]  Generating symbols
[pjproject]  Compiling apps
[pjproject]  Compiling python bindings
make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1
make[1]: *** [pjproject] Error 2
make: *** [third-party] Error 2

​I'll bet you don't have the python development libraries installed.  The 
install_prereq script was updated to include python-devel or python-dev 
depending on the distribution.​

Kind regards,
 
ROss
 
From:

[asterisk-dev] Issues with ARI python example for recording

2016-03-19 Thread Nitesh Bansal 
Hello,

I'm using the latest version of ari-py library.
I'm trying the following demo
https://wiki.asterisk.org/wiki/display/AST/ARI+and+Media%3A+Part+1+-+Recording
.

Everything is setup correctly, I can call Asterisk, but when I press the
DTMF '#' to
stop the recording, my python library throws an exception.


Entering recording state
Recording voicemail at voicemail/ 6/1458299017.29
stopping recording LiveRecording(voicemail/ 6/1458299017.29)
ERROR:ari.client:Event listener threw exception
Traceback (most recent call last):
  File "build/bdist.linux-x86_64/egg/ari/client.py", line 100, in __run
callback(msg_json, *args, **kwargs)
  File "build/bdist.linux-x86_64/egg/ari/client.py", line 198, in
extract_objects
event_cb(obj, event, *args, **kwargs)
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 181, in fn_filter
fn(objects, event, *args, **kwargs)
  File
"/root/asterisk_ari/ari-py/examples/ari_bridges/recording_demo/recording_state.py",
line 39, in on_dtmf
self.recording.stop()
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 155, in
enrich_operation
return promote(self.client, oper(**kwargs), oper.json)
  File "build/bdist.linux-x86_64/egg/ari/model.py", line 354, in promote
resp.raise_for_status()
  File "/usr/lib/python2.7/dist-packages/requests/models.py", line 773, in
raise_for_status
raise HTTPError(http_error_msg, response=self)
HTTPError: 404 Client Error: Not Found

It is throwing error on the line
'self.recording.stop'

Personally, I can't see anything wrong with this code, any ideas what I'm
doing wrong or
is there any bug in the ARI lib?

Thanks,
Nitesh
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[asterisk-dev] Building a conference application with ARI

2016-03-19 Thread Nitesh Bansal 
Hello folks,

I have just started exploring Asterisk ARI and I really like this.
I'm just wondering if anybody has built a conferencing application
with ARI?
Any examples/references are really welcome.

Many thanks,
Nitesh
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