Re: [asterisk-users] [asterisk-app-dev] ARI application execution feature survey

[Matthew Jordan]

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp  wrote:

> On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> > I get the desired use case to run app_amd from within a Stasis
> > application, but I’m not sure about app_queue. You have everything at
> > your disposal within ARI itself to replicate all of the functionality
> > of app_queue and beyond.
>
> Yes, there are certain applications which are logically building blocks to
> bigger applications. AMD is one of those which would be best if it were its
> own functionality within ARI, but allowing execution of the application is
> a good enough option. I don't think applications such as Queue, Dial,
> ConfBridge, Playback, Record or some others really make sense.
>
>
Assuming the TALK_DETECTION function isn't sufficient, it's worth noting
that the information that AMD uses to make its decisions are available to
the parts of Asterisk that make up ARI. I wonder if it would be better to
simply wrap up the existing talk detection events under some other HTTP
resource  rather than open up this entire concept.

While I'm pretty far removed from the guts of Asterisk these days, the
notion of having dialplan applications be executed from within ARI just
fills me with some fear. You can certainly open up some nightmare scenarios
where people invoke Stasis from within Stasis recursively, or invoke GoTo
or other dialplan context affecting applications.

For that matter, many of the monolithic dialplan applications have specific
options that place channels into dialplan contexts that execute after their
execution. I'm not even sure I can begin to wrap my head around what that
will do to a channel in ARI.

-- 
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Engineering
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Re: [asterisk-users] Community forum ?

[Matthew Jordan]

On Thu, Aug 30, 2018 at 3:25 PM John Covici  wrote:

> Is Sangoma taking over Digium?  Pretty soon there won't be anything
> open source around in this field at all.
>
>
Sangoma acquired Digium.

How this impacts Asterisk is answered by the community FAQ:

https://wiki.asterisk.org/wiki/display/AST/Sangoma+and+Digium+Join+Together+FAQ

tl;dr: it doesn't.




> On Thu, 30 Aug 2018 11:14:33 -0400,
> Carlos Rojas wrote:
> >
> > [1  ]
> > [1.1  ]
> > [1.2  ]
> > Is the list going to be the same after sangoma take over digium?
> >
> > On Thu, Aug 30, 2018 at 11:12 AM, Joshua Colp  wrote:
> >
> >  On Thu, Aug 30, 2018, at 12:05 PM, sean darcy wrote:
> >  > I see a lot of tag lines on posts for the Asterisk Community Forum.
> Is
> >  > that forum supposed to supersede this mailing list ?
> >
> >  Both remain available but the community forum seems to be more active,
> and it is easier to search and find things.
> >
> >  --
> >  Joshua Colp
> >  Digium, Inc. | Senior Software Developer
> >  445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> >  Check us out at: www.digium.com & www.asterisk.org
> >
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> https://community.asterisk.org/
> >
> >  New to Asterisk? Start here:
> >https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
> >  asterisk-users mailing list
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> > [2  ]
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> https://community.asterisk.org/
> >
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> >   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
> >
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>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>  John Covici wb2una
>  cov...@ccs.covici.com
>
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>
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Matthew Jordan
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Re: [asterisk-users] getting invites to rtp ports ??

[Matthew Jordan]

On Thu, Aug 30, 2018 at 6:02 AM John Covici  wrote:

> I agree, but is it possible to try over and over with anything other
> than the challenge warning in the security log as sean suggested and
> put a patch for?
>

I don't think I understand your question.

You shouldn't need a patch if you are using the SECURITY log. The thread
above is suggesting patching the source code to hijack a WARNING message
for the purposes of tracing security information; my point is that you
should have a specific SECURITY log message that already serves that
purpose.







>
> On Wed, 29 Aug 2018 22:52:05 -0400,
> Matthew Jordan wrote:
> >
> > [1  ]
> > [1.1  ]
> > [1.2  ]
> > On Wed, Aug 29, 2018 at 6:20 PM 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
> >
> > That's some pretty old advice.
> >
> > The rationale for *not* using general log messages with fail2ban still
> stands: the general WARNING/NOTICE/etc. log messages are subject to change
> between versions, and no one wants that to impact someone's security. So
> you should not use
> > those messages as input into fail2ban.
> >
> > That rationale did lead to the 'security' event type in log messages.
> Security Event Logging - as it is called - got added into Asterisk quite
> some time ago. So long ago I'm really not sure which version. At a minimum,
> Asterisk 11, but
> > I'm pretty sure it was in 10 as well.
> >
> > Documentation for it can be found here:
> >
> >
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger
> >
> > And here:
> >
> > https://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
> >
> > Note that this also fires off AMI events (and ARI events, IIRC).
> >
> > If, for whatever reason, you do not get a SECURITY log message or a
> corresponding event when something 'bad' happens, that would be worth some
> additional discussion. If anything, the events can be a bit chatty...
> >
> >
> >  -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: 1504207870-295758084-6

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

[Matthew Jordan]

munity/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
>
>
>
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>
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>
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-- 
Matthew Jordan
Digium, Inc. | CTO
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Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

[Matthew Jordan]

So, that's not quite a debug log, but just the console log with Verbose+
output.

A debug log will show a lot more information, including what the media
cache modules are trying to do when they go to get the file.

You can find information on getting debug information on the Asterisk here:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

You may also want to verify that the res_http_media_cache module is loaded.
That module is what actually does the work of pulling the remote file down
for local playback.

On Fri, Jul 20, 2018 at 3:10 PM Naftoli Gugenheim 
wrote:

> In one terminal tab:
>
> $ sudo nc -kl 80
>
> In another (note: asterisk is running in docker with --net=host):
>
> $ docker-compose exec asterisk cat /etc/hosts
> 127.0.0.1localhost
> 127.0.0.1example.com
> 127.0.1.1naftoli-ThinkPad-W540
>
> # The following lines are desirable for IPv6 capable hosts
> ::1 ip6-localhost ip6-loopback
> fe00::0 ip6-localnet
> ff00::0 ip6-mcastprefix
> ff02::1 ip6-allnodes
> ff02::2 ip6-allrouters
>
> $ docker-compose exec asterisk curl http://example.com/dummyfile.wav
> ^C⏎
>
> The HTTP request headers show up in nc.
>
> However,
>
> $ docker-compose exec asterisk asterisk -rvddT
> Seeding global EID '5c:51:4f:a5:bf:59' from 'wlp3s0' using 'siocgifhwaddr'
> Parsing /etc/asterisk/asterisk.conf
> Asterisk 15.5.0, Copyright (C) 1999 - 2016, Digium, Inc. and others.
> Created by Mark Spencer <mailto:marks...@digium.com; 
> target="_blank">marks...@digium.com>
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for 
> details.
> This is free software, with components licensed under the GNU General Public
> License version 2 and other licenses; you are welcome to redistribute it under
> certain conditions. Type 'core show license' for details.
> =
> Connected to Asterisk 15.5.0 currently running on naftoli-ThinkPad-W540 (pid 
> = 8)
> Core debug is still 6.
> [Jul 20 20:00:16] == Setting global variable 'SIPDOMAIN' to 'localhost'
> [Jul 20 20:00:16] -- Executing [1400@inbound:1] Set("PJSIP/local-004e", 
> "JITTERBUFFER(adaptive)=default") in new stack
> [Jul 20 20:00:16] -- Executing [1400@inbound:2] AGI("PJSIP/local-004e", 
> "agi://127.0.0.1/route") in new stack
> [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP learning after remote 
> address set to: 173.124.23.24:7078
> [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP qualifying stream type: audio
> [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP switching source address to 
> 127.0.0.1:7078
> [Jul 20 20:00:16] -- AGI Script Executing Application: (MixMonitor) Options: 
> (/sounds/monitor-2018-07-20T20:00:16.992040Z.wav)
> [Jul 20 20:00:16] == Begin MixMonitor Recording PJSIP/local-004e*[Jul 20 
> 20:00:16] WARNING[6384][C-0050]: file.c:772 ast_openstream_full: File 
> http://example.com/dummyfile.wav <http://example.com/dummyfile.wav> does not 
> exist in any format
> *[Jul 20 20:00:17] --  Playing 
> '/sounds/prompts/welcome-to.slin' (escape_digits=) (sample_offset 0) 
> (language 'en')
> [Jul 20 20:00:17] WARNING[6384][C-0050]: chan_iax2.c:1228 
> jb_warning_output: Resyncing the jb. last_delay 0, this delay -359631367, 
> threshold 1000, new offset 359631367
> [Jul 20 20:00:18] --  Playing 
> '/sounds/prompts/some-org.slin' (escape_digits=) (sample_offset 0) (language 
> 'en')
> [Jul 20 20:00:19] --  Playing 
> '/sounds/prompts/press-2-now-to-use-a-phone-number-other-than-the-one-you-are-calling-from-.slin'
>  (escape_digits=0123456789#*) (sample_offset 0) (language 'en')
> [Jul 20 20:00:20] WARNING[6370]: res_pjsip_registrar.c:957 
> find_registrar_aor: AOR '' not found for endpoint 'local'
> [Jul 20 20:00:21] > 0x7f9e8000cb00 -- Strict RTP learning complete - Locking 
> on source address 127.0.0.1:7078
> [Jul 20 20:00:21] -- AGI Script agi://127.0.0.1/route 
> completed, returning -1
> [Jul 20 20:00:21] == MixMonitor close filestream (mixed)
> [Jul 20 20:00:21] == End MixMonitor Recording PJSIP/local-004e
>
> Nothing shows up in nc.
>
> P.S. I have no idea why it thinks the other prompts are .slin when in
> reality they are .wav
>
> Thanks.
> ​
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Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

[Matthew Jordan]

> On Jul 20, 2018, at 1:39 PM, Naftoli Gugenheim  wrote:
> 
> I've tried it with .wav. Same result. It doesn't even hit my server.
> 

Can you provide a debug level 5 log (including all higher level verbose+ 
messages) from Asterisk that shows the playback operation?



> 
> On Fri, Jul 20, 2018, 11:45 AM Matthew Jordan  <mailto:mjor...@digium.com>> wrote:
> 
>> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim > <mailto:naftoli...@gmail.com>> wrote:
>> 
>> Crickets...
>> 
>> I've tried this now on 15.5.0. Still completely broken.
>> 
>> 
> 
> I suspect you’re encountering behavior that is working as intended.
> 
> Normally, when Asterisk plays back a file, it scans the file system for all 
> files with the provided sound file name. For each file that it finds with a 
> given file extension, it picks the best media file (where best is given by 
> transcoding cost) that matches the channel capabilities. That works great 
> when you have a file system that can be scanned quickly.
> 
> You can probably guess why that approach isn’t used with a remote HTTP 
> server: making a lot of HEAD/GET requests to ‘scan’ the remote server for 
> available file types is not a good idea for a multitude of reasons.
> 
> As such, the remote playback determines the type of file it is playing back 
> from the extension of the resource it downloads from the remote server. If 
> the remote resource doesn’t have an extension, then Asterisk is going to 
> complain that it does not know what type of media it just downloaded.
> 
> That is: if your remote resource was named “sounds/prompts/nine.wav” you’d 
> probably be okay.
> 
> Now, it would be nice if there was a way for Asterisk to be told to expect 
> the remote resource to be in a particular file format, but to my knowledge, 
> that feature hasn’t been added.
> 
> (As an aside, I use this functionality through AGI, so I know it isn’t 
> “completely broken”.)
> 
> 
> 
>> 
>> On Sun, Apr 8, 2018 at 11:28 PM Naftoli Gugenheim > <mailto:naftoli...@gmail.com>> wrote:
>> I've come back to this because of issues with the other approach I took.
>> 
>> I've set up everything so that curl http://local.XXX.com/sounds/prompts/nine 
>> <http://local.xxx.com/sounds/prompts/nine> hits my dev server, yet passing 
>> the same URL to STREAM FILE does not. I still get WARNING[103][C-0001]: 
>> file.c:774 ast_openstream_full: File 
>> http://local.mikvahbook.com/sounds/prompts/please%2Dmake%2Da%2Dselection 
>> <http://local.mikvahbook.com/sounds/prompts/please%2Dmake%2Da%2Dselection> 
>> does not exist in any format, and my server is not being hit.
>> 
>> Please help!
>> 
>> 
>> On Mon, Mar 5, 2018 at 2:49 AM Naftoli Gugenheim > <mailto:naftoli...@gmail.com>> wrote:
>> Interesting!
>> 
>> Anyway I've deployed my app, and I left it with filenames. I have a Google 
>> Cloud Storage bucket that's mounted via gcsfuse into both the app and to 
>> Asterisk. That way they both act like they're working with their own local 
>> filesystem but really it's shared but distributed. Maybe I'll change it to 
>> use URLs and serve the files from the app in the future. I feel like it's 
>> more elegant for the app to own everything and treat asterisk like a 
>> stateless service, but there's no immediate reason to change the status quo.
>> 
>> 
>> On Fri, Mar 2, 2018, 2:36 PM Ross Buggins > <mailto:rbugg...@via.co.uk>> wrote:
>> Just monitors for changes in a directory, takes the file, processes it 
>> (sends off to a web service) it and then removes it from the local file 
>> system
>> 
>>  
>> 
>> From: asterisk-app-dev-boun...@lists.digium.com 
>> <mailto:asterisk-app-dev-boun...@lists.digium.com> 
>> [mailto:asterisk-app-dev-boun...@lists.digium.com 
>> <mailto:asterisk-app-dev-boun...@lists.digium.com>] On Behalf Of Naftoli 
>> Gugenheim
>> Sent: 02 March 2018 19:30
>> 
>> 
>> To: Asterisk Application Development discussion 
>> > <mailto:asterisk-app-...@lists.digium.com>>
>> Subject: Re: [asterisk-app-dev] AGI stream audio from URI
>> 
>> 
>>  
>> 
>> How does the background service know when something was recorded?
>> 
>>  
>> 
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Re: [asterisk-users] [asterisk-app-dev] AGI stream audio from URI

[Matthew Jordan]

> On Jul 15, 2018, at 11:37 PM, Naftoli Gugenheim  wrote:
> 
> Crickets...
> 
> I've tried this now on 15.5.0. Still completely broken.
> 
> 

I suspect you’re encountering behavior that is working as intended.

Normally, when Asterisk plays back a file, it scans the file system for all 
files with the provided sound file name. For each file that it finds with a 
given file extension, it picks the best media file (where best is given by 
transcoding cost) that matches the channel capabilities. That works great when 
you have a file system that can be scanned quickly.

You can probably guess why that approach isn’t used with a remote HTTP server: 
making a lot of HEAD/GET requests to ‘scan’ the remote server for available 
file types is not a good idea for a multitude of reasons.

As such, the remote playback determines the type of file it is playing back 
from the extension of the resource it downloads from the remote server. If the 
remote resource doesn’t have an extension, then Asterisk is going to complain 
that it does not know what type of media it just downloaded.

That is: if your remote resource was named “sounds/prompts/nine.wav” you’d 
probably be okay.

Now, it would be nice if there was a way for Asterisk to be told to expect the 
remote resource to be in a particular file format, but to my knowledge, that 
feature hasn’t been added.

(As an aside, I use this functionality through AGI, so I know it isn’t 
“completely broken”.)



> 
> On Sun, Apr 8, 2018 at 11:28 PM Naftoli Gugenheim  > wrote:
> I've come back to this because of issues with the other approach I took.
> 
> I've set up everything so that curl http://local.XXX.com/sounds/prompts/nine 
>  hits my dev server, yet passing 
> the same URL to STREAM FILE does not. I still get WARNING[103][C-0001]: 
> file.c:774 ast_openstream_full: File 
> http://local.mikvahbook.com/sounds/prompts/please%2Dmake%2Da%2Dselection 
>  
> does not exist in any format, and my server is not being hit.
> 
> Please help!
> 
> 
> On Mon, Mar 5, 2018 at 2:49 AM Naftoli Gugenheim  > wrote:
> Interesting!
> 
> Anyway I've deployed my app, and I left it with filenames. I have a Google 
> Cloud Storage bucket that's mounted via gcsfuse into both the app and to 
> Asterisk. That way they both act like they're working with their own local 
> filesystem but really it's shared but distributed. Maybe I'll change it to 
> use URLs and serve the files from the app in the future. I feel like it's 
> more elegant for the app to own everything and treat asterisk like a 
> stateless service, but there's no immediate reason to change the status quo.
> 
> 
> On Fri, Mar 2, 2018, 2:36 PM Ross Buggins  > wrote:
> Just monitors for changes in a directory, takes the file, processes it (sends 
> off to a web service) it and then removes it from the local file system
> 
>  
> 
> From: asterisk-app-dev-boun...@lists.digium.com 
>  
> [mailto:asterisk-app-dev-boun...@lists.digium.com 
> ] On Behalf Of Naftoli 
> Gugenheim
> Sent: 02 March 2018 19:30
> 
> 
> To: Asterisk Application Development discussion 
> mailto:asterisk-app-...@lists.digium.com>>
> Subject: Re: [asterisk-app-dev] AGI stream audio from URI
> 
> 
>  
> 
> How does the background service know when something was recorded?
> 
>  
> 
> ___
> asterisk-app-dev mailing list
> asterisk-app-...@lists.digium.com 
> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev 
> 
> ___
> asterisk-app-dev mailing list
> asterisk-app-...@lists.digium.com
> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev

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Re: [asterisk-users] Comparison of PJSIP and SIP in Asterisk database

[Matthew Jordan]

On Tue, Mar 6, 2018 at 2:43 AM, Olivier <oza.4...@gmail.com> wrote:

> Hello,
>
> I'm currently trying to configure a passive Asterisk instance that must
> backup an active Asterisk instance.
> Each instance is connected this way:
> PSTN <---> Gateway <-- SIP --> Asterisk <-- SIP --> endpoints or IPBXs
>
> Most endpoints connect through registration.
>
> With chan_sip, Asterisk saved registration data in its database with lines
> such as:
> /SIP/Registry/spa3102 : 192.168.64.207:5060:
> 3600:7013:sip:spa3102@192.168.64.207:5060
>
> Reading such lines in active instance and copying them back in passive
> instance, I think you had a mean to have a passive instance ready to treat
> calls coming from PSTN as soon as it would become active (I never
> experimented with this).
>
> Now, with PJSIP, Asterisk saves registration data with lines such as :
> /registrar/contact/foobar: {"via_addr": ... }
>
>
>
> Have you tried to copy such registration data from one instance to an
> aother one ?
> What happened then ?
>
> Best regards
>
>
Well...

First, you should probably just use a database that is not running on the
same instance as Asterisk. You're assuming that when Asterisk dies on an
instance that it's only Asterisk that is having a problem - in more
critical failures, the AstDB (SQLite3) is going to be long gone as well. In
less critical (but still severe) failures, Asterisk will probably just be
restarted via safe_asterisk or something similar. With an external database
such as MySQL/PostreSQL, you can have one instance of Asterisk store the
registration information in the database (using Sorcery/realtime), and, if
it dies, have a spare start up and use the same database for its backing
storage. It will pick up the registration information, endpoint objects,
etc.

That being said: yes, if you can find a way to get that JSON blob from one
AstDB into another - and yes, there are ways that are sneaky but mostly
involve shenanigans and/or custom code - than a second instance of Asterisk
will understand and read that JSON just fine. Assuming it was told to get
that information from its AstDB via Sorcery as well.

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Re: [asterisk-users] PJSIP_AOR Slow

[Matthew Jordan]

On Sun, Nov 19, 2017 at 5:38 AM, Daniel Journo <d...@keshercommunications.com
> wrote:

> Hi,
>
>
>
> In my dialplans, I’m currently using PJSIP_AOR to check the status of a
> contact before dialling so that I can route the call differently if the
> endpoint is offline.
>
> But PJSIP_AOR seems to take about 0.9 seconds to return. If I’m checking
> 10 endpoints, that can cause a significant delay.
>
>
>
> Is there a better way to check the status of an endpoint pre-dialling
> within the dialplan?
>
>
>
> Here is a sample of what I’m doing.
>
>
>
> exten => example_839,9,ExecIf($["${PJSIP_AOR(example_220,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_220))
>
> exten => example_839,10,ExecIf($["${PJSIP_AOR(example_220,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,11,NoOp(${PJSIP_AOR(example_223,contact)})
>
> exten => example_839,12,ExecIf($["${PJSIP_AOR(example_223,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_223))
>
> exten => example_839,13,ExecIf($["${PJSIP_AOR(example_223,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,14,NoOp(${PJSIP_AOR(example_224,contact)})
>
> exten => example_839,15,ExecIf($["${PJSIP_AOR(example_224,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_224))
>
> exten => example_839,16,ExecIf($["${PJSIP_AOR(example_224,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,17,NoOp(${PJSIP_AOR(example_226,contact)})
>
> exten => example_839,18,ExecIf($["${PJSIP_AOR(example_226,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_226))
>
> exten => example_839,19,ExecIf($["${PJSIP_AOR(example_226,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,20,NoOp(${PJSIP_AOR(example_227,contact)})
>
> exten => example_839,21,ExecIf($["${PJSIP_AOR(example_227,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_227))
>
> exten => example_839,22,ExecIf($["${PJSIP_AOR(example_227,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,23,NoOp(${PJSIP_AOR(example_240,contact)})
>
> exten => example_839,24,ExecIf($["${PJSIP_AOR(example_240,contact)
> }"=""]?Set(UNAVAILABLEPEER=${UNAVAILABLEPEER} example_240))
>
> exten => example_839,25,ExecIf($["${PJSIP_AOR(example_240,contact)
> }"!=""]?Set(WORKINGPEERFOUND=1))
>
> exten => example_839,26,GotoIf($[${WORKINGPEERFOUND}=0]?227)
>
>
>
> Many thanks
>
> Dan
>
>
>

Where are the AORs for your endpoints stored? Static conf file, database,
etc.?

Are you using any type of caching via sorcery?

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Re: [asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

[Matthew Jordan]

location 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
>>
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>>
>
>
>
> --
> A human being should be able to change a diaper, plan an invasion, butcher
> a hog, conn a ship, design a building, write a sonnet, balance accounts,
> build a wall, set a bone, comfort the dying, take orders, give orders,
> cooperate, act alone, solve equations, analyze a new problem, pitch manure,
> program a computer, cook a tasty meal, fight efficiently, die gallantly.
> Specialization is for insects.
> ---Heinlein
>
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>
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Re: [asterisk-users] RTCP + Stasis causing high memory consumption

[Matthew Jordan]

est 13.x to
see if it resolves the issue.

If not, and you don't need the RTCP related events in either AMI or ARI,
you can permanently disable them in stasis.conf:

[declined_message_types]
decline=ast_rtp_rtcp_sent_type

decline=ast_rtp_rtcp_received_type

While that won't completely remove all processing of RTCP related
information, it will dramatically reduce the amount of work Asterisk does
when those messages are generated.

If that doesn't fix it, then you may have some form of malformed RTCP
packet that is causing Asterisk to think that it has a slew of SR/RR
reports to generate. You may want to look at the RTCP information in
wireshark to determine how many RR/SR reports are being generated in the
packets.

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Re: [asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

[Matthew Jordan]

On Wed, Oct 18, 2017 at 9:52 AM, Bryant Zimmerman <brya...@zktech.com> wrote:
> ?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip.
> We are experiencing random Jitter on outbound calls. This was not occurring
> when running asterisk 11.
>
> We have two IP's bound to pjsip one on the private vlan network the phones
> are on and the asterisk one on the asterisk wan vlan. We record the calls on
> the asterisk switch so we have the call legs. It appears that the audio is
> making it to the switch fine, but is being garbled before it leaves asterisk
> to the destination carrier. We have all media running through the server and
> this is happening when there is only 1 to 2 calls on the line. The cpu, and
> memory are not even being pushed. We are running G711 as the codec so there
> should be no real transcoding occurring..
>
> What could be causing this. The users are very upset. This is a very
> transient issue so the breakup is can occur for two to four seconds and then
> goes away. It is like asterisk and pjsip are screwing with the audio. Please
> advise.
>
> zktech

PJSIP doesn't sit in the audio stream, so that's unlikely to be the
culprit. (You've also got a lot of variables in play going from 11 =>
13 beyond just a chan_sip to chan_pjsip conversion).

Asterisk sits in the audio stream, so it could obviously be causing an
issue. Or not.

How are you recording the calls? Are you using Monitor or MixMonitor?
With what application arguments?

If you look at a packet capture, does the packet capture reveal
anything about the jitter, and on what call leg?

Have you tried using a JITTERBUFFER?

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Re: [asterisk-users] user-agent access from pjsip

[Matthew Jordan]

On Wed, Oct 18, 2017 at 11:00 AM, Bryant Zimmerman <brya...@zktech.com> wrote:
>
> I am trying to get the user-agent from extensions registered via pjsip.
> With sip we could do a sip show peer peername and it would list the 
> user-agent string.
> In a pjsip deployment it looks like this info is likely in the contact. I 
> know we can access it from the dialplan, but this is only works when a call 
> occurs. How can we get the user-agent for extensions from the console. We 
> need this for firmware version checking of extensions as many providers 
> include that in the user-agent.  Any ideas as the pjsip show contact 
> contactname does not return any real helpful info to the command line.
>
> Please advise if you are able.
>

For a long time, the project has discouraged (although not necessarily
prevented) using/abusing the CLI for interactions with external
systems. The CLI is intended for human interaction, not intersystem
interaction. Doesn't mean you can't build a system that interacts with
Asterisk through the CLI - just means you probably shouldn't.

If you need the UserAgent string for your registered endpoints, you
can get that off of the Contact. You can get the Contact via AMI by
listening for events and by querying for the status of the contacts
[1].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PJSIPShowRegistrationInboundContactStatuses

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Re: [asterisk-users] Asterisk 14 audio quality with remote files

[Matthew Jordan]

On Tue, May 16, 2017 at 3:00 PM, Tiago Ferreira
<cyberneticrevolut...@gmail.com> wrote:
> Anyone?
> I tried converting the file to g722 with ffmpeg and got the same result.
>
> regards
> Tiago
> On 12-05-2017 12:10, Tiago Ferreira wrote:
>
> Hello everyone,
>
> I am using the Asterisk REST API in order to establish a call to an endpoint
> and to send over a remote file (HTTP).
> The issue is that I am experiencing an audio quality issue.
> I have tried encoding the file differently, but everytime Asterisk is
> cutting the audio frequencies above 4Khz.
> The call is established with G.722 and the audio file is mono 16Khz 16 bit
> sln16 extension.
> What can I do to improve the sound quality? Is there any way to not have
> asterisk cut the audio frequencies?
>

The remote playback option doesn't manipulate the audio file in any
way that is different than playing the file back from disk. At a high
level, the cURL'd file is stored in a temporary location, mapped to
the URL for future referencing, and handed off to the file core. At
that point, it's the same as playing back any media file with a known
format, where that format - just like all media files in Asterisk - is
indicated by the file extension.

What extension is the file? What format does Asterisk think the file
is? It should tell you that when your verbosity is 3 or higher:

ast_verb(3, "<%s> Playing '%s.%s' (language '%s')\n",
ast_channel_name(chan), filename,
ast_format_get_name(ast_channel_writeformat(chan)), preflang ?
preflang : "default");

Matt

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Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

[Matthew Jordan]

On Thu, Feb 16, 2017 at 9:05 AM, Olivier <oza.4...@gmail.com> wrote:

>
>
> 2017-02-16 14:27 GMT+01:00 Joshua Colp <jc...@digium.com>:
>
>> On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote:
>> > Hello,
>> >
>> > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP
>> > hardphone.
>> >
>> > When a phone has enabled this feature, it would send a SIP PUBLISH to
>> its
>> > SIP Server letting this server dispatch to whatever is needs to.
>> >
>> > These messages are sent during calls but may also be sent when a call is
>> > over.
>> >
>> > At the moment, I'm using Asterisk to serve these SIP phones so my
>> > Asterisk
>> > box receives those SIP PUBLISH and discard them with a 489 Bad Event
>> > reply.
>> >
>> > I'm not using or planning to use any Kamalio server.
>> >
>> > 1. Is there an Asterisk version that would allow me to read (and store)
>> > in
>> > or out-of-band SIP PUBLISH messages from SIP phones ?
>> > 2. Alternatively, is there an Asterisk version that would allow me to
>> > relay
>> > those messages somewhere ?
>>
>> No version of Asterisk allows the handling or relaying of these
>> arbitrary PUBLISH messages. In the case of PJSIP though that is
>> pluggable so a C module could be written to do something.
>>
>
> From RFC 6035, "This document defines a new SIP event package, vq-rtcpxr,
> and
> a new MIME type, application/vq-rtcpxr, that enable the collection and
> reporting of metrics".
>
> As I'm not aware of many SIP event package currently implemented in
> PJSIP/Asterisk acting for
> out-of-calls events, it shouldn't be easy to mimic current features to add
> this new one.
>
>
>
>> > 3. Would a Kamalio-like box allow me to do this ?
>>
>> You could act as you wish on the PUBLISH requests in Kamailio.
>>
>
> This seams easier, for the moment.
>
> I think I still need to better understand what are mixed Asterisk-Kamailio
> architectures main strengths
> compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that
> is another story.
>
> Thank you very much for replying.
>
>
This admittedly high speed presentation that glosses over lots of complex
topics may or may not help you:

http://ftp.osuosl.org/pub/fosdem/2017/K.3.401/asterisk.mp4

/shameless plug off

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Re: [asterisk-users] asterisk13+app_queue scalability

[Matthew Jordan]

On Thu, Feb 2, 2017 at 3:26 AM, marek cervenka <cerva...@gmail.com> wrote:

> hi,
>
> i have similar problem to https://issues.asterisk.org/ji
> ra/browse/ASTERISK-25806
>
> do you know about some workarounds/patches for better scalability?
>
> thanks


If you've run into a situation where app_queue no longer scales for you,
you need to build your own queuing solution using Asterisk's APIs.
app_queue was not designed to scale across multiple Asterisk instances, nor
was it designed to scale up infinitely (which, of course, nothing is.)

Matt

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Re: [asterisk-users] packet loss stats - how does asterisk know about packets sent % lost ?

[Matthew Jordan]

On Sat, Jan 28, 2017 at 4:45 PM, Kevin Long <kevin.l...@haloprivacy.com>
wrote:

>
>
> Hello,
>
> I am just wondering if the statistics from the “sip show channelstats” and
> “pjsip show channelstats”  command are reliable indicators of packet loss.
> How does asterisk know how many packets *sent* were lost? Does this require
> RTCP compatible endpoint/phone,  or something else?
>
>
The number of packets lost is determined based on RTCP information received
from the far endpoint. The number is accurate so long as Asterisk is
receiving RTCP information from the endpoint(s) in question.

Matt

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Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

[Matthew Jordan]

On Mon, Nov 21, 2016 at 10:05 AM, Jonas Kellens <jonas.kell...@telenet.be>
wrote:

> On 21-11-16 15:17, Matthew Jordan wrote:
>
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens <jonas.kell...@telenet.be>
> wrote:
>
>> Hello
>>
>> when using Asterisk version 13.12.2 I notice that it takes up to 30
>> seconds (sometimes even longer) for a call queue to call its members.
>>
>> Example 1 :
>>
>> [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
>> Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
>> [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class
>> 'default', on channel 'SIP/incoming-0246'
>>
>> [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1]
>> NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
>> [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue
>> [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3]
>> Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new
>> stack
>> [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692
>>
>>
>> Example 2 :
>>
>> [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
>> Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
>> [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class
>> 'default', on channel 'SIP/incoming-0255'
>>
>> [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue
>> [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1]
>> NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
>> [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3]
>> Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new
>> stack
>> [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692
>>
>>
>> I did not see this behaviour in previous Asterisk versions.
>>
>> Could this be a bug ?
>>
>>
> There's not enough information here to know what is preventing the call
> from occurring.
>
> I'd look at a debug log between the caller entering the Queue and the
> outbound call being made. That should illustrate what is causing the delay.
>
> --
> Matthew Jordan
>
>
>
> Hello
>
>
> and what exactly am I looking for in the debug logs ?
>
> I have generated debug output and re-produced the issue.
>
>
> Again 23 seconds before calling the queue member :
>
> [Nov 21 16:23:33] pbx.c: Executing [queue@pbx-routing:15]
> Queue("SIP/incoming-4e6e", "myqueue1300,,,") in new stack
> [Nov 21 16:23:33] res_musiconhold.c: Started music on hold, class
> 'default', on channel 'SIP/incoming-4e6e'
>
> [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:1]
> NoOp("Local/mysip692@CallFromQueue-081a;2", "") in new stack
> [Nov 21 16:23:56] app_queue.c: Called Local/mysip692@CallFromQueue
> [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:2]
> NoOp("Local/mysip692@CallFromQueue-081a;2", "exten = mysip692") in
> new stack
> [Nov 21 16:23:56] pbx.c: Executing [mysip692@CallFromQueue:3]
> Dial("Local/mysip692@CallFromQueue-081a;2", "SIP/mysip692") in new
> stack
> [Nov 21 16:23:56] app_dial.c: Called SIP/mysip692
> [Nov 21 16:23:56] app_dial.c: SIP/mysip692-4e86 is ringing
> [Nov 21 16:23:56] app_queue.c: Local/mysip692@CallFromQueue-081a;1 is
> ringing
>
>
>
> Could it be that it is because my Queue member 'mysip692' is occupied in
> another bridge (call) ?
>
> This I see in the logs just before the Call Queue starts calling the queue
> member :
>
> [Nov 21 16:23:55] bridge_native_rtp.c: Locally RTP bridged
> 'SIP/mysip-4e6a' and 'SIP/incoming-4e63' in stack
> [Nov 21 16:23:55] bridge_channel.c: Channel SIP/incoming-4e63 left
> 'native_rtp' basic-bridge 
> [Nov 21 16:23:55] bridge_channel.c: Channel SIP/mysip-4e6a left
> 'native_rtp' basic-bridge 
>
>
> A bit too coincidal, no ?
>
> So then it has something to do with the bridging ?
>
>
>
> I did not have this behaviour in previous Asterisk versions.
>
>
Those aren't debug logs. Instructions for generating debug information can
be found on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

That being said, if the Queue Member is currently busy (which will be
denoted by their device state), and you have not configured the Queue to
ring the Queue Member while they are busy, then I would expect any new
caller to hang out in the Queue until that Member is available.

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Re: [asterisk-users] Asterisk 13.12.2 : strange queue behaviour

[Matthew Jordan]

On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens <jonas.kell...@telenet.be>
wrote:

> Hello
>
> when using Asterisk version 13.12.2 I notice that it takes up to 30
> seconds (sometimes even longer) for a call queue to call its members.
>
> Example 1 :
>
> [Nov 21 08:17:57] pbx.c: Executing [queue@pbx-routing:15]
> Queue("SIP/incoming-0246", "myqueue1300,,,") in new stack
> [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class
> 'default', on channel 'SIP/incoming-0246'
>
> [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:1]
> NoOp("Local/mysip692@CallFromQueue-003c;2", "") in new stack
> [Nov 21 08:18:26] app_queue.c: Called Local/mysip692@CallFromQueue
> [Nov 21 08:18:26] pbx.c: Executing [mysip692@CallFromQueue:3]
> Dial("Local/mysip692@CallFromQueue-003c;2", "SIP/mysip692") in new
> stack
> [Nov 21 08:18:26] app_dial.c: Called SIP/mysip692
>
>
> Example 2 :
>
> [Nov 21 08:20:11] pbx.c: Executing [queue@pbx-routing:15]
> Queue("SIP/incoming-0255", "myqueue1300,,,") in new stack
> [Nov 21 08:20:11] res_musiconhold.c: Started music on hold, class
> 'default', on channel 'SIP/incoming-0255'
>
> [Nov 21 08:20:45] app_queue.c: Called Local/mysip692@CallFromQueue
> [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:1]
> NoOp("Local/mysip692@CallFromQueue-0040;2", "") in new stack
> [Nov 21 08:20:45] pbx.c: Executing [mysip692@CallFromQueue:3]
> Dial("Local/mysip692@CallFromQueue-0040;2", "SIP/mysip692") in new
> stack
> [Nov 21 08:20:45] app_dial.c: Called SIP/mysip692
>
>
> I did not see this behaviour in previous Asterisk versions.
>
> Could this be a bug ?
>
>
There's not enough information here to know what is preventing the call
from occurring.

I'd look at a debug log between the caller entering the Queue and the
outbound call being made. That should illustrate what is causing the delay.

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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Matthew Jordan]

On Fri, Nov 11, 2016 at 10:46 AM, Jerry Geis <jerry.g...@gmail.com> wrote:

> >Information on timing sources can be found here:
>
> >https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
>
> >As noted on that page, ConfBridge can use any timing interface Asterisk
> >provides, and is not limited to the DAHDI timing interface. Generally,
> >timerfd is a good timing interface.
>
> >That aside, I would try to rule out external issues with the garbled audio
> >before changing the timing source. Things like:
> > - Analysis of the RTP traffic (along with potential jitter)
> > - CPU utilization with an active conference (95% idle doesn't mean that
> >some core isn't pegged)
> > - Any potential transcoding issues or codec issues
>
> >Matt
>
> Hi Matt - thanks.
>
> Looks like I am ONLY loading:
> res_timing_pthread
> res_timing_dahdi
>
> But I dont think the res_timing(x) is working on CentOS 5.
> res_timing_timerfd does not
> even seem to be compiled on this box.
>
> How do I tell for sure what its using and if its good. All I saw in the
> asterisk log was the
> two res_timing_pthread and res_timing_dadhi being loaded.
>
>
> Everything else is fine actually. It worked with the card, and withthout
> the card just sending audio to
> one endpoint has audio issues in a conference. The machine is doing
> nothign else at that time.
>
>
>
You're probably running a version of the Linux kernel that doesn't support
timerfd, hence why it isn't available.

res_timing_pthread is ... not very good. It exists as an absolute, last
ditch fall-back for Asterisk to provide a source of timing when none
exists. As such, and assuming you have ruled out all other sources of the
garbled audio, then I'm really not surprised that it isn't very effective.

Your best bet would be to:
 - Provide a hardware timing source that res_timing_dahdi can use. IIRC,
this should work even without a specific card, but does require the dahdi
kernel module to be installed and available. (I could be wrong on the need
for a physical card however, so your mileage may vary.)
 - Upgrade to a version of the kernel that res_timing_timerfd supports.
That should be Linux 2.6.26 and glibc 2.8 or later.

Personally, if I were in your shoes, I'd go with the latter. CentOS 6
should be good out of the box, and CentOS 5 is pretty long in the tooth.

Matt
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Re: [asterisk-users] Asterisk 11.24.1 garbled audio

[Matthew Jordan]

On Thu, Nov 10, 2016 at 4:00 PM, Jerry Geis <jerry.g...@gmail.com> wrote:

> I found dahdi_test...
>
>  dahdi_test
> Opened pseudo dahdi interface, measuring accuracy...
> 99.999% 99.904% 99.974% 99.814% 98.070% 97.850% 99.985% 99.887%
> 99.708% 99.899% 99.805% 99.708% 99.902% 100.000% 99.949% 99.883%
> 99.891% 99.906% 99.784% 99.719% 99.827% 99.903%
> --- Results after 22 passes ---
> Best: 100.000% -- Worst: 97.850% -- Average: 99.698465%
> Cummulative Accuracy (not per pass): 99.991
>
> seems like low numbers and not even running audio at this time.
>
> I'm thinking with the PRI card removed there is no reliable timing source.
>
> How do I get ConfBridge to have a reliable timing source?
>
>
Information on timing sources can be found here:

https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces

As noted on that page, ConfBridge can use any timing interface Asterisk
provides, and is not limited to the DAHDI timing interface. Generally,
timerfd is a good timing interface.

That aside, I would try to rule out external issues with the garbled audio
before changing the timing source. Things like:
 - Analysis of the RTP traffic (along with potential jitter)
 - CPU utilization with an active conference (95% idle doesn't mean that
some core isn't pegged)
 - Any potential transcoding issues or codec issues

Matt

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Re: [asterisk-users] SIP and RTP port and IP addresses

[Matthew Jordan]

On Thu, Nov 10, 2016 at 7:15 AM, Ethy H. Brito <ethy.br...@inexo.com.br>
wrote:

> On Thu, 10 Nov 2016 00:35:54 +0100
> Max Grobecker <max.grobec...@ml.grobecker.info> wrote:
>
> > Hi Ethy,
>
> Hi Max and All.
>
> >
> >
> > Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> >
> > > How are these parameters available from dialplan?
> > >
> > > For instance, ${SIPURI} holds the internal "IP:port" if the client is
> > > behind NAT. I need the external IP:port
> >
> >
> > You can get the peer's signalling IP address from ${CHANNEL(recvip)} and
> the
> > RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}. If you
> need
> > more information (like the codecs used) you can find other channel
> variables
> > on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+
> Function_CHANNEL
>
> H.
>
> ${CHANNEL(rtpsource)} is always returning something like "0.0.0.0:p"
> where
> p=[0-9]
>
>
You've bound to the 'bind all' address - hence why you get '0.0.0.0'. The
'p' values are the RTP port that was chosen for that call. RTP port ranges,
by default, are from 5000 to 31000.



> and ${CHANNEL(rtpdest)} returns the internal (not accessible) IP addr if
> the
> caller is behind NAT, therefore, not what I need.
>
>
The RTP destination is going to be what is negotiated in the SDP. If that's
a private IP address, then that's what you'd see there.

If you have symmetric RTP enabled, then this will switch to the address
that we are receiving RTP from. That may or may not be the original
negotiated address - if the remote end is behind a NAT, it will most likely
switch to the public IP address that we are receiving media from.




> Wouldn't these two variables have correct values only after the callee
> answers
> the call??
>
>
No. In fact, as Asterisk is a B2BUA, there are always going to be two sets
of RTP values:

 - The source/destination of the RTP stream to the inbound channel
 - The source/destination of the RTP stream to the outbound channel

The inbound channel will have its set of RTP addresses when Asterisk either
sends a Progress indication or Answers the inbound channel. The outbound
channel will have its set of RTP addresses when the far end sends a
Progress indication or Answers the outbound channel.

All of these RTP addresses may change due to:
 * NAT settings (symmetric RTP)
 * re-INVITEs, either due to Asterisk directmedia settings or re-INVITEs
initiated by the far endpoints (call hold, etc.)
 * ICE negotiation



> >
> > Please note that, if you have not disabled re-invites, the RTP address
> may
> > change while the call is running.
>
> Interesting observation.
>
> Thanx
>
> Ethy
>
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Re: [asterisk-users] Conference Call like conference done by mobile!

[Matthew Jordan]

On Wed, Oct 5, 2016 at 11:46 PM, Mandar Khire <khireman...@gmail.com> wrote:
> Hi,
> Thanks for reply.
> For use confbridge I follows link http://www.mytechrepublic.com/?p=418
> By it I manage to create Conference room & add members to it.
> But each member has to dial conference Number.
> In my scenario Only first person dial second person's number.
> Example:-
> If Person1 has 6001, Person2 6002, person3 has 6003 & so on,
> Then In confbridge as per given link example Person1 dial 1030, then person2
> dial 1030, then person3 dial 1030 & so on for conference call.
> But In my scenario Person1 dial 6002, then make it hold, then dial 6003 &
> then merge call.
> Is it depend on softphone functionality or we need to write something in
> some conf file?
> Can we do it some how?
> I tried it on mobile & I can make conference with 6 friends means total 7
> people talk to each other without dial any conference number.
>

There isn't anything in Asterisk, out of the box, that will do
*exactly* what you're describing.

You could create it, however, using ARI [1]. I'd create a special
bridge for users who dial into the system. When they're bridged with
other users, if they hit hold, I'd intercept the hold using the
HOLD_INTERCEPT [1] function, and hang up the hold initiator, keeping
the dialled party in the same bridge. When I get a new dial attempt
from the original caller, I'd put both the caller and the new callee
in the same bridge as the original callee.

This process could be repeated as many times as you want.

[1] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573
[2] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_HOLD_INTERCEPT

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Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

[Matthew Jordan]

On Wed, Sep 21, 2016 at 9:27 AM, Amit Patkar <a...@avhan.com> wrote:
> Thanks Mathew. I understand that there is no coordination between AsyncAGI &
> AMI.
> Is there any dial plan function which can tell us if there is active AMI
> session?
>

Assuming you know the client name (login name), you can use the
AMI_CLIENT [1] dialplan function to retrieve the number of sessions
they have currently established.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_AMI_CLIENT

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Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

[Matthew Jordan]

On Tue, Sep 20, 2016 at 10:49 PM, Amit Patkar <a...@avhan.com> wrote:
> It means, AMI application is no more running or crashed or lost network
> connection with asterisk server.
> In such cases call is neither answered nor disconnected by Asterisk. I want
> to detect such state and jump to next dial plan to answer or reject the
> calls
>

No, there is no automatic coordination mechanism between AsyncAGI and
AMI. In fact, AsyncAGI doesn't know *which* AMI session is even
managing the channels - it just waits for the appropriate AMI action
to come across and signal something to the channels.

Your external application would have to manage this process. A simple
solution would be to use an AMI library that supports automatic
reconnects. On a reconnect, ask Asterisk for the current channels; if
any exist, handle their recovery either by determining their
application state or by releasing them back to the dialplan.

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Re: [asterisk-users] AsyncAGI - How to jump in dial plan when no action initiated on channel or AMI user is disconnected

[Matthew Jordan]

On Sat, Sep 17, 2016 at 6:26 AM, Amit Patkar <a...@avhan.com> wrote:
> Hi
>
> Is there any way to detect inactivity on channel when AsyncAGI is used?
> I want to detect whether application handling calls using AMI & AGI has
> stopped responding.

What do you mean by "stopped responding"?

> Alternatively, how can dialplan check if there is any AMI user connected and
> decide dial plan execution?
>
> Thanks & Regards,
> Amit Patkar

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Re: [asterisk-users] Ast 13.11.2 : bridgepeer variable empty ?

[Matthew Jordan]

On Mon, Sep 19, 2016 at 8:34 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> Hello
>
> I can confirm that the variable DIALEDPEERNAME contains the information that
> I would expect in the variable BRIDGEPEER.
>
> But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of
> Asterisk version 13 ?!
>
> So if this is not the intention, then yes this is probably a bug and should
> be reported.
>

It's intentional.

The BRIDGEPEER variable is set to the parties that you are bridged
with at that moment in time. As participants enter/leave a bridge, the
BRIDGEPEER variable gets set (up to some somewhat reasonable number).
When a channel leaves a bridge, it is removed from the BRIDGEPEER
list.

You can imagine then why the BRIDGEPEER variable isn't typically set
any longer when you are in the 'h' extension - the participants all
left.

Why did this change occur?

In Asterisk 12+, all bridging in Asterisk happens using a flexible
bridging framework. That framework accommodates not just two-party
bridges, but multi-party bridges as well. In fact, all bridges can be
turned into a multi-party bridge simply by adding additional channels.
That flexibility is pretty nice, and enables some pretty interesting
features. Unfortunately, it also makes the value of BRIDGEPEER
somewhat hard to predict. It's not hard to create a scenario where the
value of BRIDGEPEER - if we didn't remove parties that left a bridge -
becomes completely arbitrary.

So what is BRIDGEPEER good for?

It's pretty useful if you're building applications on top of Asterisk
outside of the dialplan. For example, using AMI, you can query that
channel variable to get a snapshot of who all you are in a bridge with
at that point in time.

Why wasn't DIALEDPEERNAME not affected in a similar fashion?

Mostly because dialling is still 'atomic' from the perspective of the
dialplan. When Dial ends, you presumably didn't perform 10 other dials
while that application was executing. Bridging isn't that way; phones
have the ability to manipulate the bridge themselves outside of
Asterisk's control (via attended transfers).

Matt

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Re: [asterisk-users] Dial and start music on hold after timeout

[Matthew Jordan]

On Wed, Aug 24, 2016 at 6:02 AM, Israel Gottlieb <isr...@gmail.com> wrote:
> Are you sending progress?
>
>
> בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael" <vene...@gmail.com> כתב:
>>
>> I have the same exact issue. I cannot push any sounds or even Playtones to
>> the caller, unless the channel is answered, which is not possible for
>> billing reasons.
>> I am also using the Local channel & Dial(PJSIP/...).
>> I think this is a bug in Asterisk 13. The Dial function has not answered
>> yet, so the Local channel should be able to play anything to the caller,
>> without answering, in parallel with Dial.
>> Should I open a JIRA ticket?
>>

This behavior is exactly the same as it has always been. As Richard
mentioned in your other thread, there is no bug here [1]. You have
multiple options:

(1) Indicating Ringing in the dialplan. Depending on your
configuration, Asterisk will generate a 180 and pass it back to the
caller, causing them to ring or it will generate a 183 and play a
ringing tone back to the caller itself.

(2) Indicate Progress in the dialplan. This will send back a 183 to
the caller and, if possible, will send sound from Asterisk to the
caller. You then have multiple options here:
(2a) If Asterisk has the ability to perform early bridging with an
outbound channel, it will. If not, it won't - and it won't mix the
early media from multiple outbound channels.
(2b) You can play media back yourself using MoH or one of the other
sound generation applications.

(3) Wait for one of your outbound channels to pass a 180 back, and
allow that to cause the inbound channel to ring.

[1] http://lists.digium.com/pipermail/asterisk-users/2016-August/289781.html

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

[Matthew Jordan]

:27264 in sip_devicestate: Checking device state for
> peer 100
> 31 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/100 - state 5 (Unavailable)
> 32 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100' state
> '5'
> 33 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
> 9eda334cf9584d408ccd6e14eae7143a (Checking From) --From tag
> 48505f8775334f429a54d48d5b095543 --To-tag as2ad3ad05
> 34 DEBUG[-1]: chan_sip.c:25954 in handle_incoming:  Received ACK (6) -
> Command in SIP ACK
> 35 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
> '9eda334cf9584d408ccd6e14eae7143a' of Response 21834: Match Found
> 36 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
> 1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060 (Checking To) --From tag
> as212fb4c7 --To-tag 479449046
> 37 DEBUG[-1]: chan_sip.c:4200 in __sip_ack: Acked pending invite 102
> 38 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
> '1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060' of Request 102: Match
> Found
> 39 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
> checking channel drivers for SIP - 100
> 40 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 100
> 41 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/100 - state 5 (Unavailable)
> 42 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100' state
> '5'
> 43 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
> 1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060 (Checking To) --From tag
> as212fb4c7 --To-tag 479449046
> 44 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on
> '1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060' of Request 102: Match
> Found
> 45 DEBUG[-1]: chan_sip.c:20747 in handle_response_invite: SIP response 487
> to standard invite
> 46 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'ACK sip:111'
> onto UDP socket destined for 192.168.1.200:5062
> 47 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter
> for outgoing call
> 48 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
> checking channel drivers for SIP - 111
> 49 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 111
> 50 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/111 - state 1 (Not in use)
> 51 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state
> '1'
>

While it's a bit harsh, there's nothing inherently wrong with
returning a 603 in this case - so I wouldn't say it's a bug. Asterisk
has decided that since it tried everything it could about the
extension, the person at the other end must not want the call, and
hence opts for the global 6xx response as opposed to a 4xx response.

If you want to return something else, then you can provide a cause
code to the Hangup application that maps to the SIP response you'd
like to return. A mapping table exists on the wiki here:

https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause+Mappings

In your case, providing a 19 to the Hangup dialplan application should
convert that to a SIP cause of 480.

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Re: [asterisk-users] Leave and re-enter a conference

[Matthew Jordan]

On Sun, Aug 14, 2016 at 1:28 PM, Tech Support <aster...@voipbusiness.us> wrote:
> All;
>
> What I want to do is create a way to easily send callers into a
> conference room to have an N-way conference call. I created an extension
> ‘100’ that calls the MeetMe() command. Then all I need to do is transfer a
> caller using a blind transfer (*2 in my case) to extension 100. Then I can
> dial a feature code that sends me into that conference (*15 in my case). So
> far, a piece of cake. What I realize now is that once I enter the
> conference, I can’t add more people to the call. What I need is a way to
> easily exit the conference, call another user, add them to the call, etc.
> and then re-enter the room myself. I tried using the ‘p’ and ‘X’ meetme
> options without success. In other words:
>
>
>
> Place a call.
>
> Blind transfer the call to the conference (*2100)
>
> Enter the conference myself (*15)
>
> Exit the conference
>
> Repeat as necessary
>
>
>
> Any insight at all would be greatly appreciated.
>
> Thanks;
>
> John
>

This is actually where ConfBridge shines. The flexibility of
ConfBridge's menu options lets you build whatever custom actions you
want triggered from participants in the conference.

If you use the dialplan_exec DTMF menu option [1], you can have the
ConfBridge participant bounce out to the dialplan. From there, you
would execute Originate to call in another participant. Note that you
need to use Originate instead of Dial, as you would otherwise have the
participant be bridged in a new bridge with whoever they dialed.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge

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Re: [asterisk-users] PJSIP not detected

[Matthew Jordan]

On Thu, Aug 11, 2016 at 2:08 PM, Saint Michael <vene...@gmail.com> wrote:
> I installed PJSIP from the project
> git clone https://github.com/asterisk/pjproject pjproject
> cd pjproject
> make uninstall & make distclean
> ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound
> --disable-resample --disable-video --disable-opencore-amr
> --with-external-srtp
> make dep && make && make install && ldconfig && ldconfig -p | grep pj
>
> and it is there, but the configure for Asterisk 13.11.0-rc1 does not detect
> it and it cannot compile it.
> What am I doing wrong? The box is Ubuntu 14.04 LTS

Asterisk uses pkg-config to find pjproject. You can test if pkg-config
can find pjproject by running the following:

$ pkg-config --exists --print-errors libpjproject

If you don't see any error messages after that, then pkg-config is all
good. If you do see a bunch of errors, than that would explain why
Asterisk can't find pjproject.

In the case that you *do* see error messages, you'll need to inform
pkg-config of the location of the libpjproject.pc file. Some
instructions to help with that can be found on the wiki here:

https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-Troubleshooting

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

[Matthew Jordan]

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kell...@telenet.be> wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)

What in particular?

Any longer, Asterisk is *very* conservative with functionality that is
removed. Given that Asterisk 13 is simply the evolution and refinement
of the architecture introduced in Asterisk 12, I would not expect
there to be any major differences moving from 12 to 13.

> I will do so if there is no other option.
>
> But still, I don't see why Ast 13 would differ so much in this case ? If ICE
> and NAT is working (not causing problems) why should Ast 13 bring me audio
> and Ast 12 don't ??

Asterisk 13 has a lot more bug fixes than Asterisk 12. Asterisk 12 is
no longer actively supported.

Supported timelines for versions are available on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

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Re: [asterisk-users] Asterisk & Vitelity Invite issues

[Matthew Jordan]

On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly <tammy-li...@wiztech.biz> wrote:
> my bad, both sides are generating re-invites.  Vitelity ignores any
> inbound invites to continue call flow.  to keep the call going our pbx
> has to deal with their re-invites otherwise the call terminates at 30
> minutes on the dot.  Our side is ignoring the inbound invites from
> vitelity and that causes the call to be torn down.
>

The 'directmedia' or 'canreinvite' settings only apply to Asterisk
generating a re-INVITE to initiate remote packet bridging. Setting
that to 'no' will only prevent Asterisk from initiating a re-INVITE to
perform said bridging; it won't apply to anything else. There's a
whole host of reasons why Asterisk would generate a re-INVITE. That
could be due to SIP session timers, or because a change occurred in
the party identification via a connected line update. Asterisk will
generate re-INVITEs when that happens, and there isn't a setting that
will prevent that from happening.

Asterisk should have no problem accepting and handling a re-INVITE
from a provider, so long as it is formed correctly.

If your provider can't accept a re-INVITE being sent to them, there's
something seriously wrong with that provider. This is pretty core
functionality in any SIP stack.

Matt

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Re: [asterisk-users] Asterisk 13 High CPU usage

[Matthew Jordan]

On Sat, Aug 6, 2016 at 11:13 AM, Chirag Desai <djchill...@gmail.com> wrote:
> All,
>
> I upgraded to asterisk 13.10. I have minimal load on the box. 20-30 calls a
> day.
>
> Right now, there are no calls on the box at all.
>
> top shows me this:
>
> PR 20
>
> NI 0
>
> VIRT 1570540
>
> RES 84620
>
> SHR 26296
>
> S S
>
> %CPU 99.7
>
> %MEM 8.4
>
> TIME+ 3468:39
>
> COMMAND asterisk
>
> When I run this command
> while true; do top -Hbc -p `pgrep asterisk` -n 1 && asterisk -rx "core show
> threads"; sleep 1; done
>
> I get this
>
> PID USER  PR  NIVIRTRESSHR S %CPU %MEM TIME+ COMMAND
> 29079 root  20   0 1570540  84620  26296 R 37.5  8.4   1178:31 asterisk
> 29010 root  20   0 1570540  84620  26296 R 31.2  8.4   1197:07 asterisk
> 29047 root  20   0 1570540  84620  26296 R 31.2  8.4   1186:48 asterisk
>
> Any ideas??
>
>
> 
>
> Previous message
> 
>
>
> Hi all,
>
> I was using 13.5 but upgraded today to 13.9 (13.10 came out a few hours
> after I upgraded).
>
> On both 13.5 and 13.9 asterisk seems to use 100% of the CPU. This usually
> happens a few hours after starting asterisk. A restart of asterisk gets the
> CPU back down, but only for a little while.
>
> There asterisk box has no call traffic flowing through it, just 15 or so
> registrations.
>
> I'm sure this is not best practise but for now I am using chan_sip and
> pjsip at the same time. My pjsip endpoints are using TLS.
>
> I am not sure where to start looking in order to debug the CPU usage by
> asterisk and would very much appreciate some guidance.
>
> Kind regards,
>
> Chirag

Hi Chirag -

That does seem a bit odd. If you have 'core show threads', then you do
have DEBUG_THREADS enabled, which can cause a pretty hefty performance
hit - but I still wouldn't expect your CPUs to just be sitting there
spinning.

Can you get a backtrace of the threads? [1] Make sure you have
DONT_OPTIMIZE and BETTER_BACKTRACES enabled. That should show us what
the threads are doing, which would give us a better idea of what is
spending all the time processing things.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

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Re: [asterisk-users] Asterisk 13.10.0 Now Available

[Matthew Jordan]

On Thu, Jul 21, 2016 at 4:18 PM, Teijo <g.aloi...@gmail.com> wrote:
>
>
> 21.7.2016, 20:38, Asterisk Development Team kirjoitti:
>>
>> Bugs fixed in this release:
>> ---
>>  * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version.
>>   (Reported by Alexander Traud)
>
>
> Now it's possible to use dtls_cipher settings such like:
>
> dtls_cipher=ALL:!SSLv3
> or
> dtls_cipher=HIGH:!SSLv3
>
> Thank you!
>

I'll echo that sentiment - Alexander has done a lot of work recently
to improve Asterisk's support of available ciphers both in DTLS and
SRTP.

Thanks Alexander!

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Re: [asterisk-users] No Sangoma ISDN BRI cards detected by goautodial

[Matthew Jordan]

On Wed, Jul 20, 2016 at 12:14 PM, Yves biganiro <yves.bigan...@gmail.com> wrote:
> Asterisk 1.8.23.0-1_centos5.go
>
> DAHDI Version: 2.6.1 Echo Canceller: HWEC
>

I'm fairly sure that GOautodial is a packaged solution based on vicidial:

http://goautodial.org/projects/goautodialce/wiki/goautodial_getting_started_guide

As a result, you will almost certainly need to solicit help from the
GOautodial folks. Things that are packaged up in such a fashion
typically have a specialized configuration that is too specific for
the Asterisk project itself to support.

Matt

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Re: [asterisk-users] PJSIP and the pound (#) as %23

[Matthew Jordan]

On Wed, Jul 20, 2016 at 6:47 AM, Saint Michael <vene...@gmail.com> wrote:
>
> Is there any way to make PJSIP send the "#" as "#" and not as %23 in the 
> INVITE?
> I cannot figure this out.
>

The '#' character is a delimiter in URIs, and must be escaped if not
being used as such. Quoting RFC 2396, 2.4.3 [1]:

> The angle-bracket "<" and ">" and double-quote (") characters are
> excluded because they are often used as the delimiters around URI in
> text documents and protocol fields.  The character "#" is excluded
> because it is used to delimit a URI from a fragment identifier in URI
> references (Section 4). The percent character "%" is excluded because
> it is used for the encoding of escaped characters.
>
> delims  = "<" | ">" | "#" | "%" | <">

PJSIP is doing the "right thing" by escape encoding the reserved character.

[1] https://www.ietf.org/rfc/rfc2396.txt

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Re: [asterisk-users] Function SHELL not registered

[Matthew Jordan]

On Wed, Jul 6, 2016 at 4:05 AM, Michael Jepson <michael.jep...@cm.nl> wrote:
> Adding live_dangerously did the trick. Thanks! But how dangerous is Asterisk
> living now ?
>
>
>

>From README-SERIOUSLY.bestpractices.txt:

===
Avoid Privilege Escalations
===

External control protocols, such as Manager, often have the ability to get and
set channel variables; which allows the execution of dialplan functions.

Dialplan functions within Asterisk are incredibly powerful, which is wonderful
for building applications using Asterisk. But during the read or write
execution, certain diaplan functions do much more. For example, reading the
SHELL() function can execute arbitrary commands on the system Asterisk is
running on. Writing to the FILE() function can change any file that Asterisk has
write access to.

When these functions are executed from an external protocol, that execution
could result in a privilege escalation. Asterisk can inhibit the execution of
these functions, if live_dangerously in the [options] section of asterisk.conf
is set to no.

In Asterisk 12 and later, live_dangerously defaults to no.


When setting 'live_dangerously' to yes, you are taking responsibility
for preventing permission escalation for those dialplan functions that
can alter the underlying system. In addition to running Asterisk as a
non-root user - which is always a good idea - your external
applications should be sanitizing data passed through to said dialplan
functions, and should implement their own stringent access control.

Matt

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Re: [asterisk-users] VoipRaider is true for FREE calls?

[Matthew Jordan]

On Tue, May 10, 2016 at 2:42 AM, Frank Vanoni <mailingl...@linuxista.com>
wrote:

>
> On Mon, 2016-05-09 at 19:43 -0300, Vitor Mazuco wrote:
> > VoipRaider the site, says calls to landlines in Brazil...
>
> I hope I'm not infringing any mailing list rule by recommending you to
> take a look to the following providers. I use them with my Asterisk, the
> rates are good and they allow multiple calls.
>
> callwithus.com
>
> freelycall.com
>
>
While it's sometimes a grey area, discussion of commercial products and
businesses properly belongs on the asterisk-biz list:

http://lists.digium.com/mailman/listinfo/asterisk-biz

While I know conversations tend to diverge sometimes, the asterisk-users
list should be about using Asterisk, and not about promoting some third
party service or software that may pertain to Asterisk.

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Re: [asterisk-users] Dial command for SIP driver with To-header config

[Matthew Jordan]

On Fri, Apr 22, 2016 at 11:04 AM, Nitesh Bansal <nitesh.ban...@gmail.com>
wrote:

> Hello,
>
> I'm using the following Dial command syntax:
> Dial*(SIP/peer/exten!sip:x...@xyz.com <sip%3a...@xyz.com>*), the SIP URI
> after the '!' mark should be set as To-URI in outgoing INVITE
> from Asterisk.
> It works, but problem is that To-URI formatting is a bit messed up,
> It looks as follows:
> *sip:sip:x...@xyz.com <sip%3asip%3a...@xyz.com>*, it seems that Asterisk
> added an extra '*sip:'* in the
> To-header and it breaks.
>
> I'm using Asterisk 13.
> I'm wondering if this behaviour is intended or a potential bug?
>
>
I would think that it isn't a bug. If you look at the documentation of that
dial string option for the chan_sip channel driver in sip.conf.sample, you
can see that the URI scheme is left off:

  54 ; All of these dial strings specify the SIP request URI.
  55 ; In addition, you can specify a specific To: header by adding an
  56 ; exclamation mark after the dial string, like
  57 ;
  58 ; SIP/sales@mysipproxy!sa...@edvina.net

While it might be nice if it didn't always use a scheme of 'sip', that'd
probably be categorized as an improvement to this option.

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Re: [asterisk-users] CDR ODBC error

[Matthew Jordan]

On Tue, Feb 9, 2016 at 4:39 PM, Carlos Chavez <cur...@telecomabmex.com>
wrote:

> I am trying to get cdr via odbc to work on Asterisk 13.7.2 but I keep
> getting this error:
>
> [Feb  9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc:
> Error in ExecDirect: -1, query is: INSERT INTO cdr
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence)
> VALUES ({ts '2016-02-09 16:21:28'},?,?,?,?,?,?,?,?,?,?,?,?,?,?,? ,?,?,?)
> [Feb  9 16:21:43] WARNING[2088]: res_odbc.c:612 ast_odbc_direct_execute:
> SQL Execute error! Verifying connection to asterisk [asterisk]...
> [Feb  9 16:21:43] WARNING[2088]: cdr_odbc.c:160 execute_cb: cdr_odbc:
> Error in ExecDirect: -1, query is: INSERT INTO cdr
> (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield,peeraccount,linkedid,sequence)
> VALUES ({ts '2016-02-09 16:21:28'},?,?,?,?,?,?,?,?,?,?,?,?,?,?,? ,?,?,?)
> [Feb  9 16:21:43] ERROR[2088]: cdr_odbc.c:189 odbc_log: CDR direct execute
> failed
>
> First thing I do not get is that calldate does not exist in the CDR
> database (I am using the table structure that comes with the asterisk
> source).  If I add that column then start, answer and end do not get
> populated when the call ends.  Next question is which odbc cdr module I
> should use, cdr_odbc or cdr_adaptive_odbc?
>
> Also Asterisk has crashed at least three times with this message:
>
> asterisk:
> /builddir/build/BUILD/mysql-connector-odbc-5.2.5-src/driver/desc.c:110:
> desc_free_paramdata: Assertion `aprec' failed.
> [Feb  9 16:28:48] WARNING[3781]: res_odbc.c:1405 _ast_odbc_request_obj2:
> SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC
> 5.2(w) Driver]Lost connection to MySQL server during query
> [Feb  9 16:28:48] WARNING[3781]: res_config_odbc.c:117 custom_prepare: SQL
> Prepare failed![SELECT * FROM ps_domain_aliases WHERE id = ?]
> [Feb  9 16:28:48] WARNING[3781]: res_odbc.c:765 ast_odbc_sanity_check:
> Connection is down attempting to reconnect...
> Aborted (core dumped)
>

You should use cdr_adaptive_odbc. It is far more flexible than cdr_odbc,
and is essentially a replacement for it. cdr_odbc doesn't receive much
attention as a result.

Frankly, we should probably just remove cdr_odbc.

-- 
Matthew Jordan
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Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

[Matthew Jordan]

On Fri, Jan 29, 2016 at 6:15 AM, Bryant Zimmerman <brya...@zktech.com>
wrote:

> Sonny
>
> We use a real-time database for adding pjsip users. If you want to do it
> from the pjsip.conf you would have to write to the file from a script of
> some sort and then trigger a reload.   There is a real-time implementation
> for the extensions.conf as well. I personally use scripts for most of my
> dialplan, but in some cases I write to files included in my dialplan from a
> script and force a reload.
>
> To directly answer you question I do not believe there is an API baked
> into asterisk to update the pjsip.conf and extensions.conf directly from
> the dialplan.
>
> Thanks
>
> Bryant
>
> --
> *From*: "Sonny Rajagopalan" <sonny.rajagopa...@gmail.com>
> *Sent*: Thursday, January 28, 2016 7:35 PM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: [asterisk-users] Asterisk 13.6.0: Is there a way to create
> PJSIP users and dialplans programmatically using API
>
> Hi,
>
> I am using Asterisk 13.6.0 and was wondering if I can programmatically add
> users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk
> server using API of some sort.
>
> Please do let me know.
>
>
>
With the right Sorcery configuration, you can also use ARI push
configuration. Creating a PJSIP endpoint, for example, can be done with the
following:

$ curl -X PUT -H "Content-Type: application/json" -u asterisk:secret -d
'{"fields": [ { "attribute": "from_user", "value": "alice" }, {
"attribute": "allow", "value": "!all,g722,ulaw,alaw"}, {"attribute":
"ice_support", "value": "yes"}, {"attribute": "force_rport", "value":
"yes"}, {"attribute": "rewrite_contact", "value": "yes"}, {"attribute":
"rtp_symmetric", "value": "yes"}, {"attribute": "context", "value":
"default" }, {"attribute": "auth", "value": "alice" }, {"attribute":
"aors", "value": "alice"} ] }'
https://localhost:8088/ari/asterisk/config/dynamic/res_pjsip/endpoint/alice

This wiki page describes how this works, as well as how to set it up:

https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration

-- 
Matthew Jordan
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] PJSIP Returning 421 Extension Required

[Matthew Jordan]

On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard <kct...@gmail.com> wrote:

> I am turning up a PJSIP Endpoint and am having problems when they send an
> INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since
> "extension" means different things in the SIP stack versus Asterisk, I
> don't know what it is complaining about.
>
> I have attached the trace below. Nothing else shows up with core verbose
> or core debug enabled, so I am assuming it has to be dying at the PJSIP
> module. The INVITE does come from an abnormal UDP Port, which is also shown
> in the Via header, but the fact that the PBX is responding makes me think
> that isn't the culprit.
>
> Any thoughts?
>
> SIP Logger:
> INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0
> v: SIP/2.0/UDP 10.77.27.103:20065
> ;branch=z9hG4bK0020C575A392E895C39051;oc-accept
> Max-Forwards: 70
> t: <sip:+18165116504@12.4.240.200;user=phone>
> f: <sip:+18165116504@10.77.27.103;user=phone>;tag=10847511385389740959
> i: 117620342110831512016142@10.77.27.103
> CSeq: 1 INVITE
> d: no-fork
> Privacy: none
> P-Asserted-Identity:
> <sip:+18165116504;oli=62;rn=+1229218@10.77.27.103:20065;user=phone>
> Require: 100rel
> Accept: application/sdp
> k: histinfo,resource-priority
> c: application/sdp
> m: 
> Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE
> l:   228
>
> v=0
> o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55
> s=-
> c=IN IP4 10.77.160.55
> t=0 0
> m=audio 37700 RTP/AVP 0 101
> b=AS:80
> b=RR:0
> b=RS:0
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=maxptime:20
>
> <--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 --->
> SIP/2.0 421 Extension Required
> Via: SIP/2.0/UDP 10.77.27.103:20065
> ;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept
> Call-ID: 117620342110831512016142@10.77.27.103
> From: <sip:+18165116504@10.77.27.103
> ;user=phone>;tag=10847511385389740959
> To: <sip:+18165116504@12.4.240.200
> ;user=phone>;tag=z9hG4bK0020C575A392E895C39051
> CSeq: 1 INVITE
> Require: 100rel
> Supported: 100rel, timer, replaces, norefersub
> Server: Asterisk PBX 13.3.0-rc1
> Content-Length:  0
>
>
PJSIP is rejecting the inbound INVITE request as 100rel is required, but is
not in the Supported header of the inbound SIP INVITE request. I would
suspect that the UAC is doing things incorrectly by placing 100rel in the
Require but not in the list of option tags in the Supported header.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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Re: [asterisk-users] DIALSTATUS not being set

[Matthew Jordan]

On Tue, Dec 22, 2015 at 7:26 AM, Marcos Prates <mar...@ultrawave.com.br>
wrote:

> Hi,
>
> I'm having a strange problem with Asterisk 13 i can't seem to find out
> whats causing it.
> After a Dial call from one SIP peer to another, if the calling side hangs
> up, DIALSTATUS is not set, but when the called side hangs up, it does.
> The strangest thing is when debugging SIP, it sends/receives the BYE
> signal normaly on both situations.
> I'm using DIALSTATUS on my accounting/billing scripts, so when this
> happens it break the routine.
>
> Can anyone shed some light into this for me? i'm running out of ideas here.
>
> Thanks.
>
> Marcos O.
>
>
Works for me. Given the following dialplan, which has a hardcoded Dial to
PJSIP endpoint 'alice':

exten => _,1,NoOp()
 same => n,Dial(PJSIP/alice,15)
 same => n,Hangup()

exten => h,1,NoOp()
 same => n,Log(NOTICE, ${DIALSTATUS})


Calling party (bob) hangs up first:

   -- Executing [1000@default:1] NoOp("PJSIP/bob-0001", "") in new stack
-- Executing [1000@default:2] Dial("PJSIP/bob-0001",
"PJSIP/alice,15") in new stack
-- Called PJSIP/alice
-- PJSIP/alice-0002 is ringing
-- PJSIP/alice-0002 answered PJSIP/bob-0001
-- Channel PJSIP/alice-0002 joined 'simple_bridge' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
-- Channel PJSIP/bob-0001 joined 'simple_bridge' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
-- Channel PJSIP/bob-0001 left 'native_rtp' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
  == Spawn extension (default, 1000, 2) exited non-zero on
'PJSIP/bob-0001'
-- Channel PJSIP/alice-0002 left 'native_rtp' basic-bridge
<593321e8-d105-4075-94f3-20480cae3c45>
-- Executing [h@default:1] NoOp("PJSIP/bob-0001", "") in new stack
-- Executing [h@default:2] Log("PJSIP/bob-0001", "NOTICE, ANSWER")
in new stack
[Dec 22 16:32:47] NOTICE[9668][C-0001]: Ext. h:2 @ default:  ANSWER

Called party (alice) hangs up first:

*CLI> -- Executing [1000@default:1] NoOp("PJSIP/bob-", "") in
new stack
-- Executing [1000@default:2] Dial("PJSIP/bob-",
"PJSIP/alice,15") in new stack
-- Called PJSIP/alice
-- PJSIP/alice-0001 is ringing
-- PJSIP/alice-0001 answered PJSIP/bob-
-- Channel PJSIP/alice-0001 joined 'simple_bridge' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
-- Channel PJSIP/bob- joined 'simple_bridge' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
-- Channel PJSIP/alice-0001 left 'native_rtp' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
-- Channel PJSIP/bob- left 'native_rtp' basic-bridge
<0be9ceeb-014b-477c-b993-a4600ad7f9b0>
  == Spawn extension (default, 1000, 2) exited non-zero on
'PJSIP/bob-0000'
    -- Executing [h@default:1] NoOp("PJSIP/bob-", "") in new stack
-- Executing [h@default:2] Log("PJSIP/bob-", "NOTICE, ANSWER")
in new stack
[Dec 22 16:34:17] NOTICE[9740][C-]: Ext. h:2 @ default:  ANSWER


-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Asterisk CLI and database problem

[Matthew Jordan]

On Tue, Dec 22, 2015 at 3:48 AM, <pierre.guy...@orange.com> wrote:

> Hello,
>
> I have a problem related to the Asterisk CLI, when I enter "asterisk -rv",
> it cannot display the Asterisk CLI and instaed, I have this message "Unable
> to connect to remote asterisk (does /var/run/asterisk.ctl exist?)."
>
> When I check with "locate asterisk.ctl", it is indeed in the repertory
> "/var/run/".
>
> So after searching I have found we can enter "ps -A | grep asterisk" in
> order to find an asterisk "ghost" running and then kill it, and I have
> found:
>
>
>
> 1474 ?00:00:14 asterisk
> 1615 pts/300:00:13 asterisk
> 31411 ?00:00:00 safe_asterisk
> 31414 ?01:08:53 asterisk
>
>
>
> I don't know if it's the case and which one of these I should kill?
>
>
>
> Furthermore, I can access to the CLI with “asterisk –cv” but I often got
> the following warning : “db.c:288 db_execute_sql: Error executing SQL
> (COMMIT): database is locked”.
>
> Otherwise, is there another way to fix my problem?
>
> Thank you in advance!
>

I would suspect that you have installed Asterisk in such a fashion that a
particular user or user with certain permissions is required to access the
/var/run directory, as well as other directories Asterisk uses (such as
where it stores the AstDB).

You are then probably running the safe_asterisk script under a user without
sufficient permissions, and/or running/invoking the Asterisk CLI (via
"asterisk -rv") as a user with insufficient permissions.

I would double check:
(1) What user/groups own the various Asterisk directories (specified in
your asterisk.conf)
(2) What user/group you are running the safe_asterisk script under
(3) What user/group you are running as when you attempt to connect to
Asterisk


-- 
Matthew Jordan
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] asterisk 13 n-way call problem

[Matthew Jordan]

On Tue, Dec 22, 2015 at 1:47 AM, Dmitry Melekhov <d...@belkam.com> wrote:

> I spent some time reading docs and such change is not documented, so this
> is bug.
> I'll open issue...
>
>
Not necessarily. Certain aspects of features was definitely changed in 13,
and may require the use of a pre-dial handler now.

Please provide the full context of the call in Asterisk 13, including where
you set the __GOTO_ON_BLINDXFER variable. What you've included below does
not show enough information.



> 22.12.2015 10:53, Dmitry Melekhov пишет:
>
> Hello!
>
> I need to use n-way call as it described here:
>
> http://habrahabr.ru/sandbox/52259/
>
> It is in russian, but dial plan is quite clear.
> It works in asterisk 11:
>
>   -- Blind transferring OOH323/7272-6385 to '0' (context fromtransfer)
> priority 1
> -- Executing [0@fromtransfer:1] NoOp("OOH323/7272-6385", "") in new
> stack
> -- Executing [0@fromtransfer:1] NoOp("SIP/6052-0ab6", "") in new
> stack
> -- Executing [0@fromtransfer:2] Gosub("SIP/6052-0ab6",
> "dynamic-nway,6052,1") in new stack
> -- Executing [0@fromtransfer:2] Gosub("OOH323/7272-6385",
> "dynamic-nway,6052,1") in new stack
> -- Executing [6052@dynamic-nway:1] NoOp("OOH323/7272-6385", "") in
> new stack
> -- Executing [6052@dynamic-nway:1] NoOp("SIP/6052-0ab6", "") in
> new stack
> -- Executing [6052@dynamic-nway:2] Answer("OOH323/7272-6385", "") in
> new stack
> -- Executing [6052@dynamic-nway:2] Answer("SIP/6052-0ab6", "") in
> new stack
> -- Executing [6052@dynamic-nway:3] Set("OOH323/7272-6385",
> "CONFNO=6052") in new stack
> -- Executing [6052@dynamic-nway:3] Set("SIP/6052-0ab6",
> "CONFNO=6052") in new stack
> -- Executing [6052@dynamic-nway:4] Set("OOH323/7272-6385",
> "MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
> -- Executing [6052@dynamic-nway:4] Set("SIP/6052-0ab6",
> "MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
> -- Executing [6052@dynamic-nway:5] Set("OOH323/7272-6385",
> "DYNAMIC_FEATURES=") in new stack
> -- Executing [6052@dynamic-nway:5] Set("SIP/6052-0ab6",
> "DYNAMIC_FEATURES=") in new stack
> -- Executing [6052@dynamic-nway:6] MeetMe("SIP/6052-0ab6",
> "6052,1pdMXq") in new stack
> -- Executing [6052@dynamic-nway:6] MeetMe("OOH323/7272-6385",
> "6052,1pdMXq") in new stack
> -- Created MeetMe conference 1023 for conference '6052'
>   == Spawn extension (sipphones, 7272, 3) exited non-zero on
> 'SIP/6052-0ab6'
>
> As you can see both channels are passed to macro defined in
>
> __GOTO_ON_BLINDXFR=fromtransfer and everything works as expected.
>
> But I have problem
>
> I know that macros are deprecated, but, problem here is that in asterisk 13 
> GOTO_ON_BLINDXFR is executed only for one channel:
>
>
>
> -- Started music on hold, class 'default', on channel
> 'DAHDI/i1/6000-436'
> --  Playing 'pbx-transfer.ulaw' (language 'ru')
> -- Stopped music on hold on DAHDI/i1/6000-436
> -- Channel DAHDI/i1/6000-436 left 'simple_bridge' basic-bridge
> 
> -- Executing [0@fromtransfer:1] NoOp("DAHDI/i1/6000-436", "") in new
> stack
> -- Executing [0@fromtransfer:2] Gosub("DAHDI/i1/6000-436",
> "dynamic-nway,5082,1") in new stack
> -- Executing [5082@dynamic-nway:1] NoOp("DAHDI/i1/6000-436", "") in
> new stack
> -- Executing [5082@dynamic-nway:2] Answer("DAHDI/i1/6000-436", "") in
> new stack
> -- Executing [5082@dynamic-nway:3] Set("DAHDI/i1/6000-436",
> "CONFNO=5082") in new stack
> -- Executing [5082@dynamic-nway:4] Set("DAHDI/i1/6000-436",
> "MEETME_EXIT_CONTEXT=dynamic-nway-invite") in new stack
> -- Channel SIP/5082-0046 left 'simple_bridge' basic-bridge
> 
> -- Executing [5082@dynamic-nway:5] Set("DAHDI/i1/6000-436",
> "DYNAMIC_FEATURES=") in new stack
> -- Executing [5082@dynamic-nway:6] MeetMe("DAHDI/i1/6000-436",
> "5082,1pdMXq") in new stack
>   == Spawn extension (sipphonesconf, 6000, 4) exited non-zero on
> 'SIP/5082-0046'
>
>
> Is this expected or, may be, this is bug?
>
> So,as you can see, macro is not executed for Channel SIP/5082 , so this
> channel is not connected to conference.
>
> Could you tell me how can I get n-way c

Re: [asterisk-users] CEL entries over ODBC several hours late (Matthew Jordan)

[Matthew Jordan]

ntended NOT to be realtime?
> >
> > So, logically, Asterisk appears to be caching CELs to the tune of
> hundreds
> > of thousands of them at any given time - meaning if it is stopped (either
> > killed, or core stop gracefully'ed, or just "core stop now")  potentially
> > hundreds of thousands of CELs will just evaporate irretrivably.
> >
> > What can I do to mitigate this extremely slow populating of CELs over
> ODBC?
> >
> >
> Asterisk does not buffer CEL entries. If anything, it pushes the entries
> out to ODBC much more aggressively than what you would get with CDRs.
>
> An event is generated in Asterisk that corresponds to the CEL entry. That
> entry is pushed over a message bus (the 'event' message bus in 1.8 - 11;
> 'stasis' in 12+) and is picked up by the CEL core. The events are
> immediately sent to the registered backends, who also immediately write it
> out to the backend they support. In the case of ODBC, this immediately does
> an INSERT into the appropriate table.
>
> In Asterisk 1.8, you can look for a verbose level 11 message that will show
> when this occurs:
>
> ast_verb(11, "[%s]\n", ast_str_buffer(sql));
>
> In later versions, this was turned into a debug level 3 message (as
> anything over a verbose 5/debug 5 was cleaned up).
>
> If you see that message, then that will tell you when Asterisk *believes*
> it has written the CEL entry. If that doesn't show up in the database, then
> it is either in the ODBC driver or the Maria database.
>
> If you don't see that message, then something is preventing those events
> from getting delivered inside of Asterisk, which would only occur if you
> had some other serious call related issues occurring.
>
> Matt
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>



-- 
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Re: [asterisk-users] CEL entries over ODBC several hours late

[Matthew Jordan]

On Wed, Dec 9, 2015 at 6:32 AM, Stefan Viljoen <viljo...@verishare.co.za>
wrote:

> Hi guys
>
> I'm running 1.8.32.3 with CEL logging over ODBC to MariaDB 5.5.41 on the
> same Centos 7 machine.
>
> I've noticed that the CDR entries made are all in-time, e. g. the call will
> take place and the CDR entry is immediately written into the CDR table in
> the MariaDB database.
>
> However, CEL events for that CDR (which I need to process for a realtime
> display feature in my dialer software) are always several hours after the
> fact. E. g. I will make a call at 09:30, see the call immediate pop up in
> the MariaDB CDR table, but only at about 15:15 that afternoon will I see
> that call's CEL events come into the CEL table, from Asterisk I have
> examined the `show processlist` in MariaDB exhaustively to establish this
> fact.
>
> The system doesn't appear loaded, load average is about 1.1 - it's a
> quad-coare HT Intel Xeon E3-1225 with 8GB of DRAM running on an SSD for
> main
> storage.
>
> The system processes about 30 000 calls every 8 hour day, and services 90
> SIP phones.
>
> I can stop and restart the MariaDB instance for several minutes, when I
> restart it it immediately picks up on the "slow" CELs from where it was
> interrupted - more evidence that Asterisk is very slowly streaming the CELs
> through. I thought MariaDB was the bottleneck, but apparently not?
>
> If I make test inserts from a script into the CEL table, all of them
> complete so quickly a time indication does not even register for the query
> in MariaDB. Simple test queries on the CEL table are also instant, not even
> counting in the internal MariaDB query duration timer.
>
> Can anybody explain why this is that the CELs asterisk emits over ODBC are
> so delayed? Are CELs intended NOT to be realtime?
>
> So, logically, Asterisk appears to be caching CELs to the tune of hundreds
> of thousands of them at any given time - meaning if it is stopped (either
> killed, or core stop gracefully'ed, or just "core stop now")  potentially
> hundreds of thousands of CELs will just evaporate irretrivably.
>
> What can I do to mitigate this extremely slow populating of CELs over ODBC?
>
>
Asterisk does not buffer CEL entries. If anything, it pushes the entries
out to ODBC much more aggressively than what you would get with CDRs.

An event is generated in Asterisk that corresponds to the CEL entry. That
entry is pushed over a message bus (the 'event' message bus in 1.8 - 11;
'stasis' in 12+) and is picked up by the CEL core. The events are
immediately sent to the registered backends, who also immediately write it
out to the backend they support. In the case of ODBC, this immediately does
an INSERT into the appropriate table.

In Asterisk 1.8, you can look for a verbose level 11 message that will show
when this occurs:

ast_verb(11, "[%s]\n", ast_str_buffer(sql));

In later versions, this was turned into a debug level 3 message (as
anything over a verbose 5/debug 5 was cleaned up).

If you see that message, then that will tell you when Asterisk *believes*
it has written the CEL entry. If that doesn't show up in the database, then
it is either in the ODBC driver or the Maria database.

If you don't see that message, then something is preventing those events
from getting delivered inside of Asterisk, which would only occur if you
had some other serious call related issues occurring.

Matt

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Re: [asterisk-users] host parameter equivalent in pjsip.conf

[Matthew Jordan]

On Tue, Dec 8, 2015 at 10:29 AM, xaled <xa...@web.de> wrote:

> Hi,
>
>
>
> I’m trying to port our configuration form sip to pjsip channel and have
> following issue.
>
>
>
> Sip.conf has a host parameter that sets the RURI to a given value. This
> functionality is needed in some of our scenarios where we need to send
> requests to specific IP address with specific domain in RURI.
>
>
>
> I did not found an equivalent to the host parameter in pjsip
> configuration. Did I miss something?
>
>
>
> All I could come with is to get the Route header set to the needed value,
> but that does not help us in our scenarios. Below are relevant config
> settings and resulting SIP REGISTER Request.
>
>
>
> sip.conf:
>
>
>
> host=test.com
>
> outboundproxy=tcp://1.2.3.4
>
> fromuser=+12345678
>
> fromdomain=test.com
>
>
>
> REGISTER sip:test.com SIP/2.0
>
> From: <sip:+12345...@test.com>;tag=as5152122a
>
> To: <sip:+12345...@test.com>
>
> Contact: <sip:+12345678@4.3.2.1:5071;transport=TCP>
>
> User-Agent: Asterisk PBX 13.6.0
>
>
>
> pjsip.conf:
>
>
>
> client_uri = sip:+12345...@test.com
>
> server_uri = sip:test.com
>
> outbound_proxy=sip:1.2.3.4\;transport=tcp
>
>
>
> REGISTER sip:1.2.3.4;transport=tcp SIP/2.0
>
> From: <sip:+12345...@test.com>;tag=f47f3ed2-0975-4ff0-bd3b-bd5c38e594c4
>
> To: <sip:+12345...@test.com>
>
> Contact: <sip:+12345678@4.3.2.1:60938>
>
> Route: 
>
> User-Agent: Asterisk PBX 13.6.0
>
>
In order to preserve the request URI, you'll need to specify loose routing
on the SIP URI for the outbound proxy:

outbound_proxy=sip:1.2.3.4\;transport=tcp\;lr

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Re: [asterisk-users] after upgrade buttons on Dahdi phones don't work [SOLVED]

[Matthew Jordan]

On Sat, Dec 5, 2015 at 1:17 PM, Greg Woods <g...@gregandeva.net> wrote:

>
>
> On Fri, Dec 4, 2015 at 12:50 PM, Greg Woods <g...@gregandeva.net> wrote:
>
>> the first numeric button press generates a fast busy. Inbound calls to
>> the Dahdi phones work just fine.
>>
>
> I did some poking around and figured out that I could run something like
> "asterisk -r -d -d -d" and get more detailed debugging info. That produced
> this:
>
>  [Dec  5 08:16:50] DEBUG[28802][C-000b] sig_analog.c: waitfordigit
> returned '
> 8' (56), timeout = 0
> [Dec  5 08:16:50] DEBUG[28802][C-000b] sig_analog.c: Can't match 8
> from '400
> 2' in context from-internal
>
> OK, so the fast busy comes as soon as asterisk can see that there is no
> extension that starts with '8'. The dahdi-channels.conf file does declare
> from-internal as the context for the channels those phones are connected
> to. And there is no from-internal context declared anywhere. So this makes
> sense.
>
> What DOESN'T make sense is that this configuration *ever* worked. But it
> did, until the recent upgrade. The fix was to change "from-internal" to
> "internal" in dahdi-channels.conf . So that just leaves the question of how
> this configuration ever worked at all.
>
>
Sounds like you may have hit step 6...

http://plasmasturm.org/log/6debug/

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Re: [asterisk-users] endwhile jumping out of macro

[Matthew Jordan]

On Sat, Nov 28, 2015 at 7:14 AM, Ethy H. Brito <ethy.br...@inexo.com.br>
wrote:

>
> Hi
>
> I have a 3 level nested while-endwhile loop in a macro that when the
> execution reaches endwhile, it is jumping out to the While at the caller
> macro.
>
> It shouldn't since the are instructions after the endwhile.
>
> -- Executing [s@macro-call-from-outside:72]
> EndWhile("DAHDI/i1/1234567-4a7f", "") in new stack
>   == Channel 'DAHDI/i1/1234567-4a7f' jumping out of macro
> 'call-from-outside'
> -- Executing [s@macro-recurse_check_redirect_not_mailbox:7]
> While("DAHDI/i1/1234567-4a7f", "1") in new stack
>
> I checked the while-endwhile balance and it seems ok.
> I also checked if I GoTo() outside the loop. I don't.
>
> Macroexit is executed inside the while-endwhile loop in certain cases
> exiting some inner loop.
>
> Could MacroExiting inside a while loop cause this lost of balance?
>
>
Yes it could. A While loop should be terminated with an EndWhile.

Both the While application as well as the Macro application attempt to
control the PBX flow while a channel is executing within them. Terminating
an outer container of PBX flow without properly terminating an inner one
can inbalance the stack.

And just as a reminder, Macros are deprecated. They tend to have odd side
effects at times, and overly nesting Macros can result in a crash. You
should consider switching to subroutines.

Matt

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Re: [asterisk-users] endwhile jumping out of macro

[Matthew Jordan]

On Mon, Nov 30, 2015 at 11:34 AM, Ethy H. Brito <ethy.br...@inexo.com.br>
wrote:

> On Mon, 30 Nov 2015 09:40:50 -0600
> Matthew Jordan <mjor...@digium.com> wrote:
>
> > On Sat, Nov 28, 2015 at 7:14 AM, Ethy H. Brito <ethy.br...@inexo.com.br>
> > wrote:
> >
> > >
> > > Hi
> > >
> > > I have a 3 level nested while-endwhile loop in a macro that when the
> > > execution reaches endwhile, it is jumping out to the While at the
> caller
> > > macro.
> > >
> > > It shouldn't since the are instructions after the endwhile.
> > >
> > > -- Executing [s@macro-call-from-outside:72]
> > > EndWhile("DAHDI/i1/1234567-4a7f", "") in new stack
> > >   == Channel 'DAHDI/i1/1234567-4a7f' jumping out of macro
> > > 'call-from-outside'
> > > -- Executing [s@macro-recurse_check_redirect_not_mailbox:7]
> > > While("DAHDI/i1/1234567-4a7f", "1") in new stack
> > >
> > > I checked the while-endwhile balance and it seems ok.
> > > I also checked if I GoTo() outside the loop. I don't.
> > >
> > > Macroexit is executed inside the while-endwhile loop in certain cases
> > > exiting some inner loop.
> > >
> > > Could MacroExiting inside a while loop cause this lost of balance?
> > >
> > >
> > Yes it could. A While loop should be terminated with an EndWhile.
>
> I've already suspected that.
> I did some changes in the code. It is now running smooth.
>
> BTW, is there a "breakwhile" or something like that, that jumps out of a
> while-endwhile loop? Just like the "C" break command.
>
>
Yup - ExitWhile.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_ExitWhile


> >
> > Both the While application as well as the Macro application attempt to
> > control the PBX flow while a channel is executing within them.
> Terminating
> > an outer container of PBX flow without properly terminating an inner one
> > can inbalance the stack.
> >
> > And just as a reminder, Macros are deprecated. They tend to have odd side
> > effects at times, and overly nesting Macros can result in a crash. You
> > should consider switching to subroutines.
>
> Can you please point me some good tutorial on converting Macros to
> subroutines?
> Or on subroutines operation themselves?
>

The Asterisk wiki has a number of pages on this subject, including the
'special' subroutines (hangup handlers, pre-dial handlers, etc.):

https://wiki.asterisk.org/wiki/display/AST/GoSub

-- 
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Re: [asterisk-users] PJSIP and RTT in realtime

[Matthew Jordan]

On Thu, Oct 29, 2015 at 2:34 PM, Ryan, Travis <ry...@oscarwinski.com> wrote:

> So  I am using PJSIP realtime with Asterisk 13. I set the
> qualify_frequency column AORS and it now shows the RTT in milliseconds in
> the console. I want to be able to display that in a webpage, and was hoping
> the RTT would be updated in one of the realtime tables, but I don’t see it.
> The old chan_sip had this available.
>
>
>
>
Unlike chan_sip, a single table isn't used to store all the information
related to the activities happening in the stack. In this case, the round
trip time is associated with a 'contact_status' object, not the endpoint or
AoR itself (as an AoR may have multiple contacts). Unlike other sorcery
objects that typically represent configuration information, this is a
dynamic object that Asterisk typically manages transparently for you; hence
why it generally does not show up in configuration documentation. However,
since this is a sorcery object, you can specify in sorcery.conf where you'd
like that object to be persisted. Note that by default, it is persisted
using the 'memory' wizard.

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Re: [asterisk-users] pjsip show xxxx like endpoint?

[Matthew Jordan]

On Sun, Oct 18, 2015 at 12:39 PM, George Joseph
<george.jos...@fairview5.com> wrote:
> Did you open a Jira issue for this yet?  I can actually work on this this
> week.
>

I think it'd be pretty cool.

George: want me to open an issue?

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Re: [asterisk-users] Sending XML over the asterisk PJSIP

[Matthew Jordan]

On Sat, Oct 17, 2015 at 6:59 PM, Thyda ENG <ength...@gmail.com> wrote:
> I am pretty new with asterisk and actually, I want to send the image to the
> client and my process is that, first the image is uploaded to the server and
> once the image uploaded it will return the xml tag that contain the
> information about that image, Then the sip send that xml to the server,
> however I don't see any notify information on the server at all. I wonder do
> we need to config anything on the server to enable it accept the xml text ?
>

I'm going to go out on a limb and say Asterisk probably isn't going to
do what you want.

Even if Asterisk 'received' the XML in some fashion, I'm not sure what
you'd expect Asterisk to do with it. Asterisk is a media application
server; I'm not sure what it would do once it got an XML DOM that was
metadata associated with an image.

You can send arbitrary text message to/from Asterisk using SIP MESSAGE
requests. The fact that the text is XML would be immaterial to
Asterisk. That's probably the closest way to send arbitrary data to
Asterisk without writing a specific new module in the PJSIP stack.

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Re: [asterisk-users] Tim's band DEEPFALL NOT SPAM!!

[Matthew Jordan]

On Sat, Oct 17, 2015 at 10:44 AM, Tim King <tim.compnetw...@gmail.com> wrote:
> First of all I apologize for emailing everyone in one mass email like this,
> but it is the only logical way to get this done. We have restarted the
> Kickstarter campaign in hopes of raising the funds needed to get us into the
> studio with a national producer.
>
> PLEASE DONATE IF YOU CAN!
>
> No Donation is too small.
>
> It only takes 3 minutes.
>
> Here is the Link:
> https://www.kickstarter.com/projects/424887562/new-ep-music-development-30?ref=nav_search
>
> Please share this link with anyone you might know that could spare $5 toward
> a good cause.
>

I'm pretty sure this has nothing to do with the Asterisk project.

Please don't e-mail this list again with non-Asterisk related
questions or topics. Doing so will get you kicked off of the list.

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Re: [asterisk-users] Sending XML over the asterisk PJSIP

[Matthew Jordan]

On Sat, Oct 17, 2015 at 2:32 AM, Thyda ENG <ength...@gmail.com> wrote:
> Can i send XML data over the asterisk PJSIP ?
>

That's a fairly generic question. Can you be more specific about what
you are trying to accomplish?

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Re: [asterisk-users] pjsip database error when using MS SQL via ODBC

[Matthew Jordan]

On Fri, Oct 16, 2015 at 12:45 PM, Bryant Zimmerman <brya...@zktech.com> wrote:
>  I have a project that is requiring the use of MS SQL from asterisk. I get
> an error when the pjsip contact tries to update the contact table.
>
> [Oct 15 21:34:55] WARNING[3033]: res_odbc.c:649
> ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 22018:
> [FreeTDS][SQL Server]Conversion failed when converting the varchar value
> '3.00' to data type int. (101)
>
> The datatype in MySQL is integer and in MS SQL is integer. What could be the
> cause of this? Is it likely some kind of FeeTDS conversion issue?  If I
> change the MS SQL type to double the error goes away, but I am unsure of the
> long term issues associated with this.
>

Bryant:

There are two columns in the ps_contacts table that currently are
defined with an Integer type - 'qualify_timeout' and
'qualify_frequency'. Which one is currently giving the conversion
error?

Matt

Matthew Jordan
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Re: [asterisk-users] Help with voicemail

[Matthew Jordan]

On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello <lucab...@lucabert.de> wrote:
> Hi list!
>
> My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a
> voicemail.
> On two of these numbers the voicemail works without any problem, on the other
> it doesn't...
> I get this error:
>
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
> /var/spool/asterisk/voicemail/default/0039015111/unavail (format 0x100 
> (g729)): No such file or directory
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open 
> beep (format 0x100 (g729)): No such file or directory
> -- Recording the message
> -- x=0, open writing:  
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: wav, 
> 0x6edbd8
> -- x=1, open writing:  
> /var/spool/asterisk/voicemail/default/0039015111/tmp/DIqpGh format: gsm, 
> 0x7c6978
> [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a 
> codec translation path from 0x100 (g729) to 0x40 (slin)
>
> Of course, I have a
> file /var/spool/asterisk/voicemail/default/0039015111/unavail.gsm...
>
> Can someone help me to solve my problem?
>

Do you have a g729 codec module loaded? If so, does it show a
translation path between g729 and gsm? If so, do you have sufficient
encoder/decoder licenses?

Matt

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Re: [asterisk-users] Semicolon use in configuration?

[Matthew Jordan]

On Sun, Oct 11, 2015 at 8:55 PM, Juan van Rooyen
<juan.vanroo...@lightwire.co.nz> wrote:
> Hi there,
>
>
>
> Hope there is a quick answer for this.
>
> Is there an escape character in the Asterisk parser so I can use semicolon
> in asterisk configuration (specifically pjsip)?
>
>
>
> The reason I ask is that Spark NZ (previously Telecom NZ) uses BroadWorks,
> wants the Contact User to be:
>  01234567;tgrp=01234567;trunkcontext=telecom.co...@server.ip:5060;transport=udp>
>
>
>
> chan_sip never supported this, so I’m trying to get pjsip’s Contact User to
> do it by specifying the User portion.
>
> However semi-colon is treated as a comment by the Asterisk parser. Adding
> quotes (“) around the setting doesn’t seem to help.
>

Use a '\', i.e.,

contact=sip:01234567\;tgrp=01234567\;trunkcontext=...

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Re: [asterisk-users] Storing HANGUPCAUSE in CDR

[Matthew Jordan]

On Fri, Oct 9, 2015 at 8:27 AM, Ross Beer  wrote:
> Hi Andrew,
>
> Unfortunately that has stopped working when using chan_pjsip and asterisk
> 13.
>
> The CDR is closed too early after a dial attempt. This is the expected
> behaviour for Asterisk 13, however you should be able to set the variable
> before the CDR is locked/committed and before another dial attempt.
>
> The hangup_handler should be the way to do this as it's run within the same
> dial command.
>
> I think I will need to raise an issue as this has only stopped working in
> Asterisk 13.
>
> Thank you for your feedback,
>
> Ross
>

I've responded to the thread on the -dev list, as this is something
related to the CDR overhaul that occurred in Asterisk 12.

As an FYI: this isn't specific to chan_pjsip. You'd get this behaviour
regardless of the channel driver you used.

Matt

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Re: [asterisk-users] PJSIP: how to retrieve underlying SIP Call-ID

[Matthew Jordan]

On Tue, Oct 6, 2015 at 3:25 PM, Michael Ulitskiy <mulits...@acedsl.com> wrote:
> Hello,
>
>
>
> I've started to play with PJSIP and got stuck at the following problem.
>
> I need to retrieve SIP Call-ID associated with PJSIP channel.
>
> For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that
> doesn't work for
>
> outbound channel even in pre-dial or hangup handler. Whatever I do
> PJSIP_HEADER
>
> seem to be unable to read headers for outbound channel.
>
>
>
> Here's what I do:
>
>
>
> [xyz]
>
> exten => 999,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
>
> same =>
> n,Dial(PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1))
>
> exten => h,1,NoOp()
>
>
>
> [_pre_dial]
>
> exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
>
> same => n,Set(CHANNEL(hangup_handler_push)=_hangup,s,1())
>
> same => n,Return
>
>
>
> [_hangup]
>
> exten => s,1,NoOp(Call-ID: ${PJSIP_HEADER(read,Call-ID)})
>
> same => n,Return
>
>
>
>
>
> Here's the result:
>
> -- Executing [999@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "Call-ID:
> e3e249e5-7e8941dd-da386565@192.168.100.238") in new stack
>
> -- Executing [999@xyz:2] Dial("PJSIP/poly_650_2_01-006f",
> "PJSIP/xyz011101/sip:xyz011101@:5060,30,b(_pre_dial,s,1)")
> in new stack
>
> -- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) start
>
> -- Executing [s@_pre_dial:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ")
> in new stack
>
> -- Executing [s@_pre_dial:2] Set("PJSIP/xyz011101-0070",
> "CHANNEL(hangup_handler_push)=_hangup,s,1()") in new stack
>
> -- Executing [s@_pre_dial:3] Return("PJSIP/xyz011101-0070", "") in new
> stack
>
> == Spawn extension (xyz, 999, 1) exited non-zero on
> 'PJSIP/xyz011101-0070'
>
> -- PJSIP/xyz011101-0070 Internal Gosub(_pre_dial,s,1) complete
> GOSUB_RETVAL=
>
> -- Called PJSIP/xyz011101/sip:xyz011101@:5060
>
> == Using SIP RTP Audio TOS bits 184
>
> -- PJSIP/xyz011101-0070 is ringing
>
> -- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) start
>
> -- Executing [s@_hangup:1] NoOp("PJSIP/xyz011101-0070", "Call-ID: ") in
> new stack
>
> -- Executing [s@_hangup:2] Return("PJSIP/xyz011101-0070", "") in new
> stack
>
> == Spawn extension (xyz, 999, 1) exited non-zero on
> 'PJSIP/xyz011101-0070'
>
> -- PJSIP/xyz011101-0070 Internal Gosub(_hangup,s,1) complete
> GOSUB_RETVAL=
>
> == Spawn extension (xyz, 999, 2) exited non-zero on
> 'PJSIP/poly_650_2_01-006f'
>
> -- Executing [h@xyz:1] NoOp("PJSIP/poly_650_2_01-006f", "") in new stack
>
>
>
> As you can see I can get Call-ID of inbound channel, but I receive null for
> the outbound channel in both pre-dial and hangup handlers.
>
>
>
> So my question is if there's a way to retrieve SIP Call-ID for outbound
> channels?
>
> Also the 2nd question is if PJSIP_HEADER is supposed to be able to read
> headers of the outbound channel?
>

Hi Michael -

While you can use PJSIP_HEADER, the ability to retrieve the SIP
Call-ID through the CHANNEL function on a PJSIP channel was actually
just added in 13.6.0, and should be in the latest RC (13.6.0-rc2 [2]).

In either case, you're using a function as opposed to some
application, which means you do need to call the functions on the
specific channel. To get access to the outbound channel, you can use a
pre-dial handler's 'b' option [3]. The Call-ID *should* be set up on
the underlying invite session in the PJSIP dialog, even though it
hasn't been transmitted yet.

Matt

[1] https://gerrit.asterisk.org/#/c/1204/
[2] http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.6.0-rc2
[3] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers

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Re: [asterisk-users] does res_pjsip support ZRTP?

[Matthew Jordan]

On Mon, Oct 5, 2015 at 3:58 PM, Dmitriy Serov <serov@gmail.com> wrote:
> 05.10.2015 23:24, Joshua Colp пишет:
>>
>> On 15-10-05 05:22 PM, Dmitriy Serov wrote:
>>>
>>> Hello. Do I understand correctly that the current implementation
>>> res_pjsip does not support ZRTP?
>>> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html
>>
>>
>> ZRTP is not supported in Asterisk itself.
>>
>>> Nothing has changed since 2013? P.S. I greatly regret that moved from
>>> chan_sip to res_pjsip. Previously used very much lacking, and much of
>>> the promise failed. Dmitriy Serov.
>>
>>
>> Any particular examples?
>>
>
> - opus support. Ok... I know the reason why it is not supported fully this
> codec. But the existing foreign solution works fine with chan_sip and does
> not work with res_pjsip works.
> - endpoint specific ACL
> - No support for SIP message without authorization. For this reason, the
> previously working functionality of sending and receiving SMS from gateway
> GOIP had to rewrite their internal Protocol.
> - found hardphones and software phones that don't accept "long nonce" and
> refuse to register when using res_pjsip
> - enable icesupport also leads to problems of registration and cannot be
> "common solution"
> - issue tracker now contains multiple error messages that arise every day
> and reboot my server (which cannot be called a production)
> - And watchdog logs SegFaults and Hangs including other stacks that are not
> yet documented in the issue tracker.
>
> Be sure to have forgotten something, because it is not documented all meet
> and unsolved problems,workarounds.
>
> The transition to PJSIP was chosen as mainstream and full support for
> WebRTC. As a result, instead of developing a service I a few months I'm
> returning opportunities to which users are accustomed and expect to see.
> Having the knowledge and the overall picture a few months ago I would not
> have taken such a decision.
>

I know this is shocking to hear, but this is an open source project.

That means anyone can fix something. Anyone can add something. Even
you! You have all the power to affect your system.

It also means that no one is under any obligation to do it for you.

Surprising, right? I know, it's amazing to think that *YOU* have all
the responsibility and power.

We use PJSIP. We use it in a variety of settings. It works well for
us. Does that mean it works well for you? I don't know. I'm not you. I
don't have your use cases. Would I like it to work well for you? Of
course! But if you don't participate by reporting issues, testing
changes, and contributing code, there's not much I can do for you
other than to note that the line is long, and feel free to stand in it
until someone in the community gets around to what you'd like to have
done.

Matt

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Re: [asterisk-users] Respond to an out of call SIP MESSAGE

[Matthew Jordan]

On Mon, Sep 28, 2015 at 11:38 AM, Emil Ohlsson  wrote:
> Ah, so I can use
>
> MessageSend(sip:alice)
>
> to send a message to Alice then (reusing the existing TLS session). That does 
> seem to work. Thanks :-). I didn't know you could use users there.
>
> Is there a variable or some other method to see which user that did send the 
> message? I'm thinking something in the lines of
>
> [context]
> exten => _X!,1,NoOp(Handling message from ${SENDER})
>
> I didn't see any useful information using dumpchan, so I'm guessing there 
> isn't any variable for it. $CALLERID(name) didn't contain the name and 
> $SIP_HEADER seems to be focused on calls.
>
> Since the information is in the SIP header it should be possible to get.
>

Since MESSAGE requests are serviced on a single, special channel that
is not a SIP channel, chan_sip specific functions/variables will not
work on that channel.

The MESSAGE_DATA function will read headers off of a received SIP
MESSAGE request:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE_DATA

You can also add headers to an outbound SIP MESSAGE request using that
same function.

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Re: [asterisk-users] Respond to an out of call SIP MESSAGE

[Matthew Jordan]

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <da...@vex.net> wrote:
> On Mon, 21 Sep 2015 06:48:52 +
> Emil Ohlsson <e...@svep.se> wrote:
>> [sip-im]
>> exten _X!, 1, NoOp(Got message)
>> exten _X!, n, Answer()
>> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...)
>> exten _X!, n, SendText(Message received)
>
> I am not an expert but perhaps you want this.
>
> [sip-im]
>   exten s,1,NoOp(Got message)
> same,n,Answer()
> same,n,Agi(agi://localhost/messagehandler.agi?...)
> same,n,SendText(Message received)
>
> Replacing "exten _X!" with "same" is just a shortcut.  I find that
> there are lots of places where spaces cause problems so I just remove
> them all for good measure.  Finally, I am not sure what the mechanism
> is here but if it is like a goto then I think that you want the 's'
> priority.
>
> Or, I totally don't know what I am talking about and my education will
> be advanced by the replies to this message.  :-)
>

If you want to send an out of call SIP MESSAGE request, you'll need to
use the MessageSend application:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Application_MessageSend

SendText is used for sending text messages within a call. Since a SIP
channel is not servicing the out of call text message, you cannot use
it to send a SIP MESSAGE request back to whatever sent the original
SIP MESSAGE request.

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Re: [asterisk-users] res_pjsip. Turn off the authorization request for an incoming MESSAGE

[Matthew Jordan]

On Mon, Sep 7, 2015 at 3:24 AM, Dmitriy Serov <serov@gmail.com> wrote:
>
> Hello.
> Continue a months-long struggle that is associated with the transfer from 
> chan_sip to res_pjsi p. Where are many gates (GSM gate) that do not support 
> authentication when sending MESSAGE. For example, 4goip when relay incoming 
> SMS. Using chan_sip it was not a problem. Using res_pjsip is the problem :( 
> Is any way to turn off the authorization request for an incoming MESSAGE 
> using res_pjsip? Or any workaround? [2015-09-07 06:01:14] DEBUG[12947] pjsip: 
> sip_endpoint.c Processing incoming message: Request msg MESSAGE/cseq=542 
> (rdata0x7f88642fdc28) [2015-09-07 06:01:14] VERBOSE[12947] 
> res_pjsip_logger.c: <--- Received SIP request (447 bytes) from 
> UDP:109.165.111.xx:5807 ---> MESSAGE sip:sm...@85.142.148.xx SIP/2.0 Via: 
> SIP/2.0/UDP 109.165.111.xx:5807;branch=z9hG4bK837973400 Route: 
> <sip:85.142.148.xx;lr> From: <sip:srv_918588x...@85.142.148.xx>;tag=284759743 
> To: <sip:sm...@85.142.148.xx> Call-ID: 76603@192.168.1.100 CSeq: 542 
> MESSAGE Contact: <sip:srv_918588x...@109.165.111.xx:5807> Max-Forwards: 30 
> User-Agent: dble Content-Type: text/plain Content-Length: 35 111 Ваш баланс 
> 68,08 rub. [2015-09-07 06:01:14] DEBUG[23059] pjsip: sip_endpoint.c 
> Distributing rdata to modules: Request msg MESSAGE/cseq=542 
> (rdata0x7f88640a9288) [2015-09-07 06:01:14] DEBUG[23059] 
> res_pjsip_endpoint_identifier_ip.c: No identify sections to match against 
> [2015-09-07 06:01:14] DEBUG[23059] res_pjsip_endpoint_identifier_user.c: 
> Retrieved endpoint srv_9185880046 [2015-09-07 06:01:14] DEBUG[23059] pjsip: 
> endpoint .Response msg 401/MESSAGE/cseq=542 (tdta0x7f88717063b0) created 
> [2015-09-07 06:01:14] VERBOSE[23059] res_pjsip_logger.c: <--- Transmitting 
> SIP response (479 bytes) to UDP:109.165.111.xx:5807 ---> SIP/2.0 401 
> Unauthorized Via: SIP/2.0/UDP 
> 109.165.111.xx:5807;rport=5807;received=109.165.111.xx;branch=z9hG4bK837973400
>  Call-ID: 76603@192.168.1.100 From: 
> <sip:srv_918588x...@85.142.148.xx>;tag=284759743 To: 
> <sip:sm...@85.142.148.xx>;tag=z9hG4bK837973400 CSeq: 542 MESSAGE 
> WWW-Authenticate: Digest 
> realm="ruvoip.net",nonce="1441594874/5741fb37496404a4aa5cf0e53a129867",opaque="7441b8c64eddc67a",algorithm=md5,qop="auth"
>  Server: ruVoIP.net PBX Content-Length: 0


Your endpoint, ' srv_9185880046', most like has an auth object
specified for it. If it did not, then the MESSAGE request would not be
challenged. If you know that requests for that endpoint should not be
authenticated, then you can remove the auth option from the endpoint
and it should allow the request to proceed without a 401 challenge
response.

If you need to authenticate certain requests while allowing others
through, then today, there is no way to accomplish that in the PJSIP
stack. As an open source project, someone could certainly propose that
functionality if they wanted.

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Re: [asterisk-users] Problem with Cisco CUBE when dialling into Asterisk 13 server

[Matthew Jordan]

On Tue, Sep 1, 2015 at 2:02 AM, Brendan Ord <b...@staff.onthenet.com.au> wrote:
> Hello,
>
>
>
> This is a problem with my Cisco CUBE (2811), so apologies for this being
> kind of off-topic.  It is acting as a border for my Asterisk 13 server
> though J
>
>
>
> Rather than re-type the details of my problems, I have a post in the Cisco
> community with running-configs and various debugs attached.  I’m drawing
> blanks as to my problem so I am reaching out wherever I can to try resolve
> this.
>
>
>
> https://supportforums.cisco.com/discussion/12589596/cisco-ube-hangs-calls-immediately-after-being-answered
>

I'm not sure anyone on here is going to be able to help you, unless
they are intimately familiar with Cisco CUBE as well. Looking at the
post you referenced, where 172.22.4.8 is Asterisk, you have the
following call flow:

U 172.22.4.12:59803 -> 172.22.4.8:5061
INVITE sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.8:5061 -> 172.22.4.12:5060
SIP/2.0 100 Trying.

U 172.22.4.8:5061 -> 172.22.4.12:5060
SIP/2.0 180 Ringing.

U 172.22.4.8:5061 -> 172.22.4.12:5060
SIP/2.0 180 Ringing.

### Pick up handset to answer
U 172.22.4.8:5061 -> 172.22.4.12:5060
SIP/2.0 200 OK.

U 172.22.4.12:59803 -> 172.22.4.8:5061
BYE sip:61756767463@172.22.4.8:5061 SIP/2.0.

U 172.22.4.8:5061 -> 172.22.4.12:5060
SIP/2.0 200 OK.


If 172.22.4.12 - which I assume is the Cisco phone or CUBE - has
decided to send Asterisk a BYE, there's not much anyone can tell you
unless they are familiar with that device. Asterisk is being told to
hang up the call, and so it will do so.

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Re: [asterisk-users] No audio when using TLS/SRTP with Kamailio and Asterisk 13

[Matthew Jordan]

On Tue, Aug 18, 2015 at 3:12 AM, Chirag Desai djchill...@gmail.com wrote:

 Hi all,

 I'm using Kamailio + Asterisk 13 (PJSIP), where Kamailio (using rtpengine)
 acts as the registrar and forwards all calls to Asterisk.

 This works fine when using udp / tcp and RTP. When switching to TLS/SRTP,
 the call is set up correctly, however, I get no audio.

 When I skip kamailio and connect my two endpoints to asterisk directly I
 get a perfect call with SRTP.

 The same is also true when I skip asterisk and have the call handled by
 Kamailio (using RTPEngine).

 In PJSIP my transports look like this:

 [transport-tcp]
 type=transport
 protocol=tcp;udp,tcp,tls,ws,wss
 bind=0.0.0.0:5060
 local_net=[asterisk local ip]/17
 external_media_address=[asterisk external ip]
 external_signaling_address=[asterisk external ip]

 [transport-tls]
 type=transport
 protocol=tls
 bind=0.0.0.0:5063
 ca_list_file=/etc/asterisk/certificates/cert.crt
 cert_file=/etc/asterisk/certificates/certificate.crt
 priv_key_file=/etc/asterisk/certificates/key.key
 method=tlsv1


 My endpoint looks like this:

 [kamailio]
 type=endpoint
 context=kam_out
 disallow=all
 allow=alaw
 allow=g722
 allow=ulaw
 allow=gsm
 aors=kamailio
 direct_media=no
 media_encryption=sdes
 media_address=[Asterisk Local IP]
 rtp_symmetric=yes
 force_rport=no
 rewrite_contact=yes
 outbound_proxy=sip:[Kamailio Local IP]:5060\;transport=tcp\;lr

 [kamailio]
 type=identify
 endpoint=kamailio
 match=[Kamailio Local IP]/17

 [kamailio]
 type=aor
 contact=sip:[Kamailio Local IP]:5060\;transport=tcp


 My dialplan looks like this

 [kam_out]

 exten = 1001,1,Playback(demo-echotest)  ; Let them know what's going on
 same = n,Echo ; Do the echo test
 same = n,Playback(demo-echodone)  ; Let them know it's over
 same = n,Hangup()


 exten = _kb-.,1,NoOp(Calling a registred user with number ${EXTEN})
 same = n,Set(callee=${PJSIP_HEADER(read,To)})
 same = n,Set(callee=${callee:5})
 same = n,Set(callee=${callee:0:-1}) ; removes the 
 same = n,Dial(PJSIP/kamailio/sip:${callee})
 same = n,Hangup()

 When a call comes via kamailio it comes with a prefix of 'kb' if the value
 is an extension e.g. 1000 - 1999. Otherwise users can dial a prefix of 45
 e.g. 451001 to hit the Echo Test.

 As mentioned the echo test works fine, however the actual call between two
 endpoints has no audio. RTP debug shows nothing. PJSIP shows two channels
 in a simple bridge, but no sound. Usually PJSIP says RTP Probation passed
 and shows the IP address but in this case it does not.


The PJSIP stack only provides SIP signalling; it doesn't interfere with the
media handling in Asterisk. The handling of media is done by the RTP engine
implementation, res_rtp_asterisk.

I don't think this is a problem, however, with res_rtp_asterisk or
Asterisk. If RTP debug doesn't show any traffic, then Asterisk is almost
certainly not receiving any media.

What does a PCAP show? I'd look at where the RTPEngine is forwarding your
RTP packets off to, and see if they are getting sent somewhere other than
Asterisk.



 I'm guessing the issue is something funny in PJSIP, although I'm not 100%
 since it does work when I turn SRTP and TLS off.

 For testing I'm using CsipSimple and a Snom 760. Both are set with SRTP
 mandatory and are using TLS to talk to Kamailio.

 When kamailio talks to asterisk it uses TCP over a local network.

 I've been pulling my hair out for days. I really would appreciate any
 ideas or some pointing in the right direction here.

 Thanks in advance,

 C

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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk (Stefan Viljoen)

[Matthew Jordan]

On Mon, Aug 17, 2015 at 2:01 AM, Stefan Viljoen viljo...@verishare.co.za
wrote:

 Hi List

 Regarding this Asterisk instance as discussed previously (Asterisk
 1.8.11.0)
 that was consuming enormous amounts of file descriptors (100 000+ for about
 50 simultaneous calls) it appears I have managed to solve my problem by
 upgrading the 1.8.11.0 Asterisk instance to an 1.8.32.3 Asterisk instance.

 Also, the file descriptors apparently leaking were paired with timer
 problems in 1.8.11.0 whenever I went above about 50 concurrent calls on the
 box while running on 1.8.11.0.

 The thing is in our setup we have about 15 instances of 1.8.11.0 at the
 various branches of the company, all running 1.8.11.0, BUT at none of these
 sites do we ever exceed 40 simultaneous calls.

 The defining factor was (in our case, with our dialplan) to run 1.8.11.0
 and
 try to run 50+ concurrent calls.

 What would happen was that thousands of these messages would come up in the
 CLI:

 [Aug 13 09:41:38] ERROR[25193]: res_timing_dahdi.c:89 dahdi_timer_set_rate:
 Failed to configure DAHDI timing fd for 0 sample timer ticks

 when we reached or exceeded 50 calls.

 The same happened whether pthread timing or kernel timerfd timing was used.

 Several other weird errors would manifest in the CLI, to whit:

 ---
 format_gsm.c:102 gsm_write: Bad write (32/33): Destination address required

 [Aug 12 12:23:33] WARNING[29436]: channel.c:1474 __ast_queue_frame:
 Exceptionally long voice queue length queuing to Local/number@local-3E1C;1

 WARNING[8210]: res_rtp_asterisk.c:1773 ast_rtcp_read: RTCP Read error: Bad
 file descriptor.  Hanging up.

 [Aug 12 09:56:55] WARNING[29931]: file.c:198 ast_writestream: Translated
 frame write failed
 ---

 when we were spamming the timer errors. Practical effects were dropped
 calls, calls with bad quality / what sounds like severe jitter, and
 mixmonitor recording files that were not written or corrupt.

 The solution (so far, still checking) was simply to upgrade to 1.8.32.0 and
 most of our problems disappeared, for us in our setup, with our dialplans.
 The upgrade was painless, since we stayed in the 1.8 range, we did not have
 to modify any of our config files or dialplans.

 Maybe this can assist someone else struggling with older 1.8 series timer
 issues.

 Regards


Always nice to hear that we fixed things. Thanks for the follow-up!


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Re: [asterisk-users] Siren7 for Asterisk 13.5

[Matthew Jordan]

On Mon, Aug 10, 2015 at 10:38 AM, Richard Kenner ken...@gnat.com wrote:
 A Siren codec is not currently available and the one for 12 will not
 work. I have no timeframe for when this might change.

 So the only option is to build one from the Polycom sources?  I'm
 already doing this for Siren14 (I forget why).


Alas, until we get off our butts, yes. Sorry about that.

Really, we're putting as much effort into fixing things and issues
that affect a lot of people. While siren7/siren14/silk are nice, there
aren't as many people using them as other affected things at this
moment.

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Re: [asterisk-users] Modifying CDR values from a hangup extension in Asterisk 13

[Matthew Jordan]

 a patch
ready that modifies the behaviour, feel free to post it for review on
Gerrit as well [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Code+Review




 On 08/03/2015 04:36 PM, jg wrote:


 I'm trying to migrate from Asterisk 1.8 to Asterisk 13 and can't figure
 this one out. I'm pretty sure the question has been already asked, but I
 failed to find a solution.

 Can you modify CDR values in an h-extension?

 My cdr.conf contains:
 [general]
 enable=yes
 unanswered=yes
 endbeforehexten=yes
 initiatedseconds=no
 batch=no

 The diaplan contains a simple h extension
 exten = h,1,NoOp(${CDR(userfield)})
 exten = h,n,Set(CDR(userfield)=changed)
 exten = h,n,NoOp(${CDR(userfield)})

 In the same context I execute:
 exten = 10,1,Set(CDR(userfield)=empty)
 exten = 10,n,Dial(SIP/10)

 The h extension outputs two lines with userfield set to empty. I would
 expect the second one to be changed. It seems that I can read the CDR
 values, but I can't change them. Is it a bug or a design thing? Am I
 missing something?

 I am not working with h-extensions myself, but the docs (
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_cdr
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_cdr)
 say something like this:

 endbeforehexten
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_cdr#Asterisk13Configuration_cdr-general_endbeforehexten

 Boolean

 1

 false

 Don't produce CDRs while executing hangup logic

 This would indicate that at least writing is disabled.

 jg




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Re: [asterisk-users] Asterisk 11.19.0 Now Available

[Matthew Jordan]

On Sat, Aug 8, 2015 at 8:26 AM, Administrator TOOTAI ad...@tootai.net wrote:
 Le 07/08/2015 23:54, Asterisk Development Team a écrit :

 The Asterisk Development Team has announced the release of Asterisk
 11.19.0.

 [...]

 Hello,

 We have problem with patches since 11.18.0 We have to download the full
 tar.gz to get last version :-(.

 Before this, since ages, we used to patch the previous version like

 #patch -p0  ../asterisk-11.17.0-patch

 (applied to the current asterisk-11.16-0 directory), compile and install.
 That's all, servers where uptodate, job done.

 Taking a look at the header from asterisk-11.17.0-patch (and previous) we
 see

 --- asterisk-11.16.0-summary.html  (.../11.16.0)   (revision 433916)
 +++ asterisk-11.16.0-summary.html  (.../11.17.0)   (revision 433916)

 which is, diff between asterisk-11.16.0 and -in this case- the new
 asterisk-11.17.0

 Now, since 11.18.0 version, patch is looking like

 diff --git a/.version b/.version
 index cde331b..3644f46 100644
 --- a/.version
 +++ b/.version
 @@ -1 +1 @@
 -11.19.0-rc1
 \ No newline at end of file
 +11.19.0

 \ No newline at end of file

 As you can see patch is build against 11.19.0-rc1, not 11.18.0

 How can we apply this patch to a legacy asterisk-11.18.0 tar.gz ?

 Thanks for any hint.


That's a bug in the release scripts, which had to be rewritten when we
moved to Git. We'll try to get it sorted out for the next release.

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Filters

[Matthew Jordan]

On Mon, Jul 27, 2015 at 4:51 AM, Stefan Viljoen viljo...@verishare.co.za
wrote:

 Hi list

 I'm using Asterisk 1.8.11.0 - is there any way to apply (for example) a
 bandpass filter to Asterisk RTP audio in the realtime audio stream?

 I'm looking for a way to (for example) filter out a 50Hz AC hum present in
 some calls I push through my asterisk.

 Thanks


If you're willing to write C, then yes, what you're looking to do is
possible.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Messages out of calls. Is it really possible?

[Matthew Jordan]

On Fri, Jul 10, 2015 at 11:51 AM, Rodrigo Pimenta Carvalho
pime...@inatel.br wrote:

 Hi.

 I have read in some web sites that ASTERISK can support messages out of 
 calls. What does it exactly means?

 1 - Can a dialplan script accept and handle a message from a callee party, 
 even before the call be connected?

Since it is out of call, yes.

SIP MESSAGE requests are handled by the respective channel driver
(chan_sip or the res_pjsip stack) and passed to the dialplan using a
special hidden channel, Message. That channel caries the payload and
some meta information about the MESSAGE request, which can be accessed
using the generic out-of-call messaging functions [1].

Likewise, you can send an out of call SIP MESSAGE request using MessageSend [2].

Note that all of this has been supported since Asterisk 10.

 2 - Can a ringing callee send SIP MESSAGE to the ASTERISK even before answer 
 the call?

Yes, hence the term out-of-call.

 3- Could I use dialplan function MESSAGE() to receive SIP messages from 
 callees, even before the call be connected?

It does not receive messages; it accesses data on the message
currently being serviced by the executing Message channel.

chan_sip/res_pjsip will receive and dispatch MESSAGE requests at any
point in time. They have nothing to do with your normal SIP or PJSIP
channels, and hence nothing to do with whatever INVITE request derived
channels are currently executing in the dialplan.

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE
and https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE_DATA
[2] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend

-- 
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Can't install gmime22

[Matthew Jordan]

On Thu, Jul 9, 2015 at 1:18 PM, Harel Cohen ha...@mayorcom.com wrote:
 No one could assist?
 Could someone please tell me on which repository I can find Gmime22-devel
 for 64-bit Centos6.5?
 Is gmime-devel good or do I need to have gmime22-devel?
 What will happen if I don't install gmime22?
 Thank you...
 Harel

 Message: 3
 Date: Mon, 6 Jul 2015 02:53:51 +0200
 From: Harel Cohen ha...@mayorcom.com
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Can't install gmime22
 Message-ID: 00a601d0b786$3a7bffd0$af73ff70$@mayorcom.com
 Content-Type: text/plain;   charset=us-ascii

 Hello list,
 I'm trying to install gmime22 package which is one of the packages reported
 as required by ./contrib/scripts/install_prereq test.
 Whatever I do I'm getting to a dead end.
 On the regular yum repositories that I use (centos, epel, rpmforge,
 asterisk, digium) it is not found.
 I've found it on Fedora repositories however trying to use those I get all
 sorts of errors:

 On fedora17 repository:
 ERROR You need to update rpm to handle:
 rpmlib(X-CheckUnifiedSystemdir) is needed by filesystem-3-2.fc17.x86_64
 When I try to update rpm I'm getting a conflict between some systemd package
 to kernel

 On fedora21 repository:
 Not found

 On fedora20 repository it is reported as installed but with these errors:
 Error unpacking rpm package filesystem-3.2-19.fc20.x86_64
 error: unpacking of archive failed on file /bin: cpio: rename
   Cleanup: glibc-common-2.12-1.149.el6_6.9.x86_64
 7/10
   Cleanup: bash-4.1.2-29.el6.x86_64
 8/10
 Non-fatal POSTUN scriptlet failure in rpm package bash
 warning: %postun(bash-4.1.2-29.el6.x86_64) scriptlet failed, exit status 127
   Cleanup: glibc-2.12-1.149.el6_6.9.x86_64
 9/10
 warning: /etc/localtime saved as /etc/localtime.rpmsave
 ...and also the system hang on shutdown and won't boot again
 Could you please advise how to properly install this package?

 I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit

 Thank you...

gmime is only required for the res_http_post module. If you don't need
that module, you really don't need that dependency.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels

[Matthew Jordan]

On Fri, Jul 3, 2015 at 1:46 PM, Alonso Genis abge...@gmail.com wrote:
 Hello,

 I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with 
 success.

 I have an application that sends an action Originate to AMI for
 calling, it's working well, but when i see to Asterisk's CLI, i see 2
 calls for just one originate:

 pftestes40copiabh*CLI core show channels verbose
 Channel  Context  ExtensionPrio State
  Application  Data  CallerIDDuration
 Accountcode PeerAccount BridgeID
 SIP/1903-00091903_aux 1 Up
  AppDial  (Outgoing Line)   190300:00:12 1902
   19027866921b-4675-4823-8
 Local/1902@1902_in-0 macro-atende s1008
 Ringing Dial SIP/1902,30,t 1903
 00:00:15 19029428460a-f4e7-46d1-b
 Local/1902@1902_in-0 macro-atende s1008 Up
  Dial SIP/1903,30,t 190200:00:15 1902
   19027866921b-4675-4823-8
 SIP/1902-00081902_aux 1 Up
  AppDial  (Outgoing Line)   190200:00:15 1902
   9428460a-f4e7-46d1-b
 4 active channels
 2 active calls

 In fact, just one call is up.

 Somebody knows if this is ok, or it's a bug? May be someday asterisk
 will create just one call for one originate?

 Thanks in advanced for your answers!

It isn't a bug.

The output of 'core show channels' reports a 'call' (which is not a
concept that is represented well anywhere in Asterisk) as a channel
with a PBX thread running. In this case, that's the two channels in
your output that are not outbound channels, i.e., the Local channels
that dialled your SIP channels.

That fact that you have two different SIP channels means that
something either performed two Originates, or you have done a parallel
Dial.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Distributed Device States - Best Option

[Matthew Jordan]

On Sat, Jun 27, 2015 at 11:28 AM, Bryant Zimmerman brya...@zktech.com wrote:
 We have used AIS for disturbed Device State in the past, BLF and MWI, We are
 in the process of an update on one of our clustered systems, We are looking
 at XMPP and I found a few discussions on a Corosync with has OpenAIS built
 in.

 My question is which should I be looking at to replace my current AIS option
 I currently have.  XMPP or Corosync?

 It looks like the Corosync is just the AIS option more nicely packaged. Is
 XMPP a better solution as I grow my network? Are there down sides to XMPP
 that AIS/Corosync does better...

 Can anyone recommend where I can find some up to date documentation that
 would cover up through Asterisk 13 on Distributed Device State.


I'd take the following as opinion, and not gospel. Most of the work
I've done setting up distributed device state in Asterisk has been
either for development or testing; for production anecdotes, you'd
probably want someone else's opinion.

Both Corosync and XMPP functionally work the same. That is, from the
perspective of Asterisk, there really isn't any difference. The
question than is one of deployment.

XMPP is rather easy to set up, but does require an XMPP server. This
introduces another component, and another point of failure, into your
system. If the XMPP server goes down, your Asterisk instances will
stop aggregating device state.

Corosync is generally harder to set up, but since it is a library used
by Asterisk on your system, there isn't another discrete component
that you have to maintain and run. The Asterisk instances themselves
are set up as a cluster, which means they are generally more aware
of the other instances existence.

All of that being said, there is a third option: use SIP.

Asterisk 13's PJSIP stack also has the ability to PUBLISH device state
and MWI information between Asterisk instances. The benefit of this is
- if you are using the PJSIP stack - there is no additional components
in Asterisk to configure beyond what you are already setting up.

More information on distributing device state and MWI can be found on the wiki:

XMPP: 
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub
Corosync: https://wiki.asterisk.org/wiki/display/AST/Corosync
PJSIP: 
https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP


-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] error trying to get PJSIP working

[Matthew Jordan]

On Thu, Jun 18, 2015 at 1:52 PM, Ryan, Travis ry...@oscarwinski.com wrote:
 I’m doing an upgrade from Asterisk 11 to 13. I’m following the guide at
 https://wiki.asterisk.org/wiki/display/ast/setting+up+PJSIP+REaltime to
 setup realtime, as I use realtime on Asterisk 11 too.



 I’m getting the following error when trying to connect the peer to the
 server.



 Help? J



 Thanks,



 Travis



 [Jun 15 16:20:03] NOTICE[5116] res_odbc.c: res_odbc: Connected to laf [laf]

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] ERROR[5116] res_pjsip_registrar.c: Unable to bind contact
 'sip:812@10.1.80.112:5062' to AOR '812'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM
 ps_contacts WHERE id LIKE ? ORDER BY id

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') =
 '812;@%'

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: SELECT * FROM
 ps_contacts WHERE id LIKE ? ORDER BY id

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id LIKE') =
 '812;@%'

 [Jun 15 16:20:03] DEBUG[5116] res_odbc.c: odbc_release_obj2(0x7f3f1c4815d8)
 called (obj-txf = (nil))

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Skip: 0; SQL: INSERT INTO
 ps_contacts (id, outbound_proxy, expiration_time, path, qualify_frequency,
 user_agent, uri) VALUES (?, ?, ?, ?, ?, ?, ?)

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 1 ('id') =
 '812;@sip:812@10.1.80.112:5062'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 2
 ('outbound_proxy') = ''

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 3
 ('expiration_time') = '1434399723'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 4 ('path') = ''

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 5
 ('qualify_frequency') = '0'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 6 ('user_agent')
 = 'Media5-fone/4.1.3.3034 iOS/8.3'

 [Jun 15 16:20:03] DEBUG[5116] res_config_odbc.c: Parameter 7 ('uri') =
 'sip:812@10.1.80.112:5062'

 [Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute returned an error
 -1: 42S22: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1]Unknown
 column 'outbound_proxy' in 'field list' (103)

 [Jun 15 16:20:03] WARNING[5116] res_odbc.c: SQL Execute error -1! Verifying
 connection to laf [laf]...


It looks like you are missing the outbound_proxy column on the
ps_contacts table. If you're missing that column, you are probably
missing some other columns as well.

Note that the schema for the realtime tables for PJSIP has been
updated many times, as new features have been added. The alembic
scripts bundled with Asterisk can manage your DB schemas for you, or
can be used to generate the schema used by your specific version of
Asterisk.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Calling multiple phones at ones

[Matthew Jordan]

 Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Demkovitch
 Sent: Sunday, June 14, 2015 7:13 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Calling multiple phones at ones

 Hello group!

 I’m new to Asterisk but got one running finally :)

 Now I’m trying to solve following problem. I have company Automated Attendant 
 and each employee have
 SIP phone at home, SIP phone in office, cell phone.

 I want all those 3 phones to be “one”. So, if someone calls our company 
 number and dials my extension - I’d like 3 phones to ring at the same time.

 What is this feature and where should I look for samples, etc? I’m going by 
 “Asterisk: The definite guide” book and pretty confident with those concepts 
 described but not sure
 how to achieve what I described above.

 Thank you,
 Ivan
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Re: [asterisk-users] queue periodic-announce without stopping ringing

[Matthew Jordan]

On Mon, Jun 15, 2015 at 9:22 AM, Marek Cervenka cerv...@fpf.slu.cz wrote:
 hello,

 is it possible to play queue periodic-announce without stopping agents
 ringing? actual situation is sequential - ring agents, play announce (for 15
 sec), ring agents , ... (i need to connect agent with caller asap when agent
 is free)

 is it possible with ARI?


ARI does not change or otherwise allow for the manipulation of the
mechanics in app_queue. If you want to use app_queue, ARI is not the
API for you.

If you are looking to write your own call queue application, than ARI
has the ability to manipulate media on a channel, as well as whether
or not ringing is being indicated to the channel. Since you want to
ring an agent, play media to the agent, then ring the agent again, you
will most likely need to indicate ringing to the agent using inband
ringing (via a 'tone' media URI [1]).

For more information on ARI and its intended use, see [2].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Channels+REST+API#Asterisk13ChannelsRESTAPI-play
[2] https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573

Matt

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Re: [asterisk-users] ARI echo test

[Matthew Jordan]

On Fri, May 22, 2015 at 4:41 AM, Nick Awesome jl...@me.com wrote:
 Can anyone tell me how can I create echo test using ARI stasis application?


I'm not sure an 'echo' test really makes much sense with ARI, but we
do have some nice documentation on getting started with ARI on the
wiki. The basic tutorial example should give you an ARI event over a
WebSocket connection.

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

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Re: [asterisk-users] DPMA - Asterisk Realtime

[Matthew Jordan]

On Fri, May 1, 2015 at 10:43 AM, Robert Broyles rob...@webservicesaz.com
wrote:

 We love our Digium phones and DPMA - but we really need it to work on our
 Realtime Platform. Otherwise we lose all the cool features and they are
 just standard SIP phones.

 Anyone working on a solution for this? Or anyone from Digium see this on
 the roadmap?


Hey Robert -

We've had a number of requests to have the DPMA work more closely with
Asterisk Realtime. Right now, that feature isn't planned for an upcoming
scheduled release, but we do keep track of requests such as this. We've
made a note of it, and we'll keep evaluating it versus other planned and
requested features.

Thanks -

Matt

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Re: [asterisk-users] Asterisk 1.8.32.3 chan_sip deadlock

[Matthew Jordan]

On Wed, Apr 29, 2015 at 8:42 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hello asterisk-users,

 We've been having intermittent issues with chan_sip - it stops responding to
 cli requests, trying to reload chan_sip from cli doesn't seem to have any
 effect, initiated calls carry on for a short period, but no new SIP requests
 are processed ('sip show channels' hangs forever, server stops responding to
 SIP OPTIONS, or any other SIP messages). We have updated the build from
 1.8.23.1 to the latest asterisk 1.8 (1.8.32.3), however the problem still
 persists. We have gathered debugging information from 'core show locks' and
 from gdb, attached to this message (with phone numbers and extension and
 context names obscured). We are running realtime under CentOS 6.6, built
 from source and packaged using rpmbuild, with the following menuselect
 options (debugging version):
 menuselect/menuselect --disable BUILD_NATIVE --enable DEBUG_THREADS --enable
 DONT_OPTIMIZE --disable CORE-SOUNDS-EN-GSM --disable-category
 MENUSELECT_EXTRA_SOUNDS --disable MOH-OPSOUND-WAV --enable-category
 MENUSELECT_ADDONS --disable format_mp3 --disable cdr_tds --disable cel_tds
 --disable cdr_pgsql --disable cel_pgsql --disable res_config_pgsql
 menuselect.makeopts

 under kernel 2.6.32-504.el6.x86_64, and linked against the following library
 versions:

 /usr/lib64/libssl.so.10:symbolic link to `libssl.so.1.0.1e'
 /usr/lib64/libcrypto.so.10: symbolic link to `libcrypto.so.1.0.1e'
 /lib64/libc.so.6:   symbolic link to `libc-2.12.so'
 /usr/lib64/libxml2.so.2:symbolic link to `libxml2.so.2.7.6'
 /lib64/libz.so.1:   symbolic link to `libz.so.1.2.3'
 /lib64/libm.so.6:   symbolic link to `libm-2.12.so'
 /lib64/libdl.so.2:  symbolic link to `libdl-2.12.so'
 /lib64/libpthread.so.0: symbolic link to `libpthread-2.12.so'
 /lib64/libtinfo.so.5:   symbolic link to `libtinfo.so.5.7'
 /lib64/libresolv.so.2:  symbolic link to `libresolv-2.12.so'
 /lib64/libgssapi_krb5.so.2: symbolic link to `libgssapi_krb5.so.2.2'
 /lib64/libkrb5.so.3:symbolic link to `libkrb5.so.3.3'
 /lib64/libcom_err.so.2: symbolic link to `libcom_err.so.2.1'
 /lib64/libk5crypto.so.3:symbolic link to `libk5crypto.so.3.1'
 /lib64/libkrb5support.so.0: symbolic link to `libkrb5support.so.0.1'
 /lib64/libkeyutils.so.1:symbolic link to `libkeyutils.so.1.3'


 We'd appreciate any possible assistance, as we're having problems working
 out what exactly triggers the deadlock and we have not been able to find the
 correct sequence of steps to reproduce the issue yet, other than waiting for
 it to lock up at an arbitrary time with the debugging code in place. It does
 seem to happen at least once a day, however.

 What is the best way of getting the core show locks output for people to see
 as it appears to be too big to mail?


Please go ahead and make an issue on the issue tracker. Make sure you
get both the output of 'core show locks', as well as a GDB backtrace.
Instructions for both can be found here:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

-- 
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Re: [asterisk-users] Asterisk proxying a REFER

[Matthew Jordan]

On Mon, Apr 27, 2015 at 10:36 AM, Luca Pradovera
luca.pradov...@gmail.com wrote:
 Hello,
 we are using Asterisk with Adhearsion as our application server, with
 another Asterisk box acting as the office PBX, where all office phones are
 registered.

 A REFER to transfer calls within the office results in the Adhearsion
 application call being dropped, because the leg between the PBX and the app
 server is terminated by the PBX following the REFER.
  Is there a way to configure Asterisk 11 to proxy a refer across a bridge
 instead of following it, so the application server can follow it instead?


Hey Luca -

Unfortunately, there is not a simple or easy configuration setting
that tells Asterisk to proxy the REFER request through. Generally,
Asterisk doesn't like proxying anything.

There may still be another way to handle this issue, depending on the
setup. Can you provide a bit more information about the channels on
the PBX/Adhearsion server, who sends the REFER request, and what
happens explicitly in the scenario?

Matt

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Asterisk 11 SRTP: unsupported crypto parameters: UNENCRYPTED_SRTCP

[Matthew Jordan]

On Fri, Apr 17, 2015 at 6:16 AM, Satish Barot satish4aster...@gmail.com wrote:
 Hi All,

 I have Asterisk 11 talking to Avaya over SIP trunk using TLS and SRTP.
 On incoming calls from Avaya asterisk complains of 'unsupported crypto
 parameters: UNENCRYPTED_SRTCP' and rejects the call with '488 Not acceptable
 here'

 Doesn't Asterisk support  UNENCRYPTED_SRTCP as crypto parameters in sdp?

 FYI SDP looks like this.

 v=0
 o=- 1429194215 1 IN IP4 XX.XX.XX.XX
 s=-
 c=IN IP4 XX.XX.XX.XX
 b=TIAS:64000
 t=0 0
 a=avf:avc=n prio=n
 a=csup:avf-v0
 m=audio 50096 RTP/SAVP 0 18 120
 a=rtpmap:0 PCMU/8000
 a=rtpmap:18 G729/8000
 a=fmtp:18 annexb=no
 a=rtpmap:120 telephone-event/8000
 a=ptime:20
 a=crypto:1 AES_CM_128_HMAC_SHA1_80
 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP

 And on CLI I see,

 DEBUG[1568][C-] sip/sdp_crypto.c: local_key64
 7vXot5kn/sl/GYv5ENN6yW0PZZapQ00c++biLgoX len 40
 WARNING[1568][C-] sip/sdp_crypto.c: Unsupported crypto parameters:
 UNENCRYPTED_SRTCP
 DEBUG[1568][C-] chan_sip.c: Processing media-level (audio) SDP
 a=crypto:1 AES_CM_128_HMAC_SHA1_80
 inline:zUVSWsFB/WjVtLxXojBT7zbNvuQ4BkOwcCkD/AjM|2^20 UNENCRYPTED_SRTCP...
 UNSUPPORTED OR FAILED.
 WARNING[1568][C-] chan_sip.c: Rejecting secure audio stream without
 encryption details: audio 50096 RTP/SAVP 0 18 120
 VERBOSE[1568][C-] chan_sip.c:
 --- Reliably Transmitting (NAT) to XX.XX.XX.XX:5061 ---
 SIP/2.0 488 Not acceptable here

 Thanking in advance for any inputs.


Asterisk is complaining because placing an UNENCRYPTED_SRTCP after
the lifetime parameter in a crypto attribute is part of RFC 4568
(Security Descriptions for Media Streams), which Asterisk does not
support.

You will need to see if the Avaya system can be configured to not send
the attribute.

-- 
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Re: [asterisk-users] meetme vs confbridge max user comparison wanted

[Matthew Jordan]

On Mon, Apr 13, 2015 at 1:15 PM, Steve Edwards
asterisk@sedwards.com wrote:
 I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme
 and I'd like to switch to confbridge to service more callers.

 Can anyone reply with their experience along the lines of 'using meetme I
 was only getting x callers per server but with confbridge I now get y
 callers per server?'


Anecdotally, when ConfBridge was first rewritten in Asterisk 10, some
performance comparisons with MeetMe were performed. In the best case,
on a particular system with conference user usage patterns, we saw
MeetMe hit a limit at around 60 channels, and ConfBridge reach over
240 channels. Worst case for ConfBridge was around 140 channels.

Note that the ConfBridge sample rate, mixing interval, and other
parameters can greatly affect how far it scales out.

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[asterisk-users] Asterisk is moving to Git

[Matthew Jordan]

Hello!

For quite some time now, there's been a desire in the Asterisk project
to move the project source control from Subversion to Git. After a lot
of work and planning, we believe we are finally able to start that
process.

Starting on Monday, April 13th, the Asterisk project's Subversion
repository will be set to read-only. New changes will no longer be
made in any of the Subversion branches. A new Git repository for the
Asterisk project will be set up using Gerrit [1] as the primary
repository, with mirrors conveniently located at git.asterisk.org [2].

While we've done a lot of work to plan for this migration, things can
(and probably will) happen during this process. As such, there may be
some hiccups during the next week or two while we iron out the finer
points that such a large change will have on the project. If you are
used to pulling directly from the project's Subversion repository,
please be patient as we make this leap. At the same time, if you do
happen to encounter any issues, please don't hesitate to send an
e-mail to the asterisk-dev mailing list [3] or talk with the
developers in the #asterisk-dev IRC channel.

As always, thanks for supporting the Asterisk project!

Matt

[1] https://gerrit.asterisk.org
[2] https://git.asterisk.org
[3] http://lists.digium.com/mailman/listinfo/asterisk-dev

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Re: [asterisk-users] Asterisk 13.3.0 Centos Package Install Error

[Matthew Jordan]

On Mon, Apr 6, 2015 at 6:19 AM, Aaron Hunter aaron.hunt...@gmail.com wrote:
 I'm trying to install 13.3.0 from binary packages and get an error. It
 worked for 13.2.0 but not for 13.3.0.

 Centos 6.6 64bit fresh install

 Followed instructions on the wiki:
 rpm -Uvh
 http://packages.asterisk.org/centos/6/current/i386/RPMS/asterisknow-version-3.0.1-2_centos6.noarch.rpm
 yum update
 yum install asterisk asterisk-configs --enablerepo=asterisk-13

 Yum output:
 [goes through the usual stuff but then...]

 -- Finished Dependency Resolution
 Error: Package: asterisk-core-13.3.0-1_centos6.x86_64 (asterisk-13)
Requires: libilbccodec.so.2()(64bit)
Available: pjproject-2.1-0.digium2.1_centos6.x86_64
 (asterisk-current)
libilbccodec.so.2()(64bit)
Available: pjproject-2.3-0.digium2.1_centos6.x86_64
 (asterisk-current)
libilbccodec.so.2()(64bit)
Available: pjproject-2.3-0.digium3.1_centos6.x86_64
 (asterisk-current)
libilbccodec.so.2()(64bit)
Available: pjproject-2.1-0.digium1.1_centos6.x86_64
 (asterisk-current)
Not found
Available: pjproject-2.1-0.digium1.2_centos6.x86_64
 (asterisk-current)
Not found
Available: pjproject-2.3-5.el6.i686 (epel)
Not found
 Error: Package: asterisk-core-13.3.0-1_centos6.x86_64 (asterisk-13)
Requires: libg7221codec.so.2()(64bit)
Available: pjproject-2.1-0.digium2.1_centos6.x86_64
 (asterisk-current)
libg7221codec.so.2()(64bit)
Available: pjproject-2.3-0.digium2.1_centos6.x86_64
 (asterisk-current)
libg7221codec.so.2()(64bit)
Available: pjproject-2.3-0.digium3.1_centos6.x86_64
 (asterisk-current)
libg7221codec.so.2()(64bit)
Available: pjproject-2.1-0.digium1.1_centos6.x86_64
 (asterisk-current)
Not found
Available: pjproject-2.1-0.digium1.2_centos6.x86_64
 (asterisk-current)
Not found
Available: pjproject-2.3-5.el6.i686 (epel)
Not found


 Does anyone have any idea what might be wrong?


I just ran this on a CentOS 6.6 64-bit VM and couldn't reproduce the
issue you're seeing.

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Re: [asterisk-users] CDR dst value null after attended transfer

[Matthew Jordan]

|
 +-+++--++--+--+-+--+---+
 | 2015-03-26 12:11:04 | 1427382664.963 | 1427382664.963 |  645 |
 5491549116 | 7051 |   27 | ANSWERED| SIP/pabx-e1-0252 |
 SIP/7051-0253 |
 | 2015-03-26 12:11:32 | 1427382664.963 | 1427382664.963 |  649 |
 5491549116 |  |7 | ANSWERED| SIP/pabx-e1-0252 |
 SIP/7003-0255 |
 +-+++--++--+--+-+--+---+
 2 rows in set (0.62 sec)

 Notice how the dst field on the second line is missing.

 Am I doing something wrong here or this is a bug?


Looks like you're hitting this bug:

https://issues.asterisk.org/jira/browse/ASTERISK-24443


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Re: [asterisk-users] Dial to PJSIP Channel with Typo PJSIP// Causes Asterisk Shutdown

[Matthew Jordan]

On Thu, Mar 26, 2015 at 9:28 AM, Trey Hilyard kct...@gmail.com wrote:
 I found an issue with how PJSIP handles a typo in the Dial application. If
 the Channel is mistakenly typed with two slashes (i.e Dial(PJSIP//...),
 the Dial applications fails (obviously), but it also kills the server.

 I put some code in my pbx_config to check for that string and not let the
 dialplan reload, but it seems like there should be a better way to handle in
 in the PJSIP stack or Dial app so that it doesn't take the server down if it
 gets through.

 I am not a developer, but I was hoping maybe someone who monitors this
 mailing list might feel like taking this on as a bug fix.I haven't tried
 with any other channel drivers, so it may cross to others.


Please open an issue on the issue tracker:

https://issues.asterisk.org/jira

A backtrace from the crash will be needed as well. Instructions on
generating a backtrace can be found on the wiki here:

https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

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Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34

[Matthew Jordan]

On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
 hello list,

 i have asterisk 11.15.0 and i have some trunks sip from my provider

 we have some ip phone astra 6731i

 each Ip-phone is configured with trunk and we call

 no ihave configured another trunk from the same provider in my asterisk

 i can call all numbers just the numbers are configured in thses ip phones.

 but when i configured the same trunk in x-lite i can call theses ip-phones
 without issue
  the problem just when i configure the trunk in my server and i use
 extension

 all the ip-phone and x-lite and server asterisk in the same network
 192.168.1.x

  == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called SIP/FD/0033149XX
 -- SIP/FD-00b9 is making progress passing it to SIP/306-00b8
 0x2afec424c430 -- Probation passed - setting RTP source address to
 192.168.1.212:57592
 0xc5922b0 -- Probation passed - setting RTP source address to
 217.195.xx.xxx:29674
 -- Got SIP response 556 No address found back from 217.195.XX.XXX:5060
   == Everyone is busy/congested at this time (1:0/1/0)
 -- Executing [s@macro-dialout-trunk:23] NoOp(SIP/306-00b8, Dial
 failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34)
 in new stack
 -- Executing [s@macro-dialout-trunk:24] GotoIf(SIP/306-00b8,
 0?continue,1:s-CONGESTION,1) in new stack
 -- Goto (macro-dialout-trunk,s-CONGESTION,1)
 -- Executing [s-CONGESTION@macro-dialout-trunk:1]
 Set(SIP/306-00b8, RC=34) in new stack
 -- Executing [s-CONGESTION@macro-dialout-trunk:2]
 Goto(SIP/306-00b8, 34,1) in new stack
 -- Goto (macro-dialout-trunk,34,1)
 -- Executing [34@macro-dialout-trunk:1] Goto(SIP/306-00b8,
 continue,1) in new stack
 -- Goto (macro-dialout-trunk,continue,1)
 -- Executing [continue@macro-dialout-trunk:1] NoOp(SIP/306-00b8,
 TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
 other trunks) in new stack
 -- Executing [continue@macro-dialout-trunk:2] Set(SIP/306-00b8,
 CALLERID(number)=306) in new stack
 -- Executing [0149XX@from-internal:7] Macro(SIP/306-00b8,
 outisbusy,) in new stack
 -- Executing [s@macro-outisbusy:1] Progress(SIP/306-00b8, ) in
 new stack
 -- Executing [s@macro-outisbusy:2] GotoIf(SIP/306-00b8,
 0?emergency,1) in new stack
 -- Executing [s@macro-outisbusy:3] GotoIf(SIP/306-00b8,
 0?intracompany,1) in new stack
 -- Executing [s@macro-outisbusy:4] Playback(SIP/306-00b8,
 all-circuits-busy-nowpls-try-call-later, noanswer) in new stack
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
 ast_openstream_full: File all-circuits-busy-now does not exist in any format
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
 ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
 such file or directory
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
 playback_exec: ast_streamfile failed on SIP/306-00b8 for
 all-circuits-busy-nowpls-try-call-later, noanswer
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
 ast_openstream_full: File pls-try-call-later does not exist in any format
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
 ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such
 file or directory
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
 playback_exec: ast_streamfile failed on SIP/306-00b8 for
 all-circuits-busy-nowpls-try-call-later, noanswer
 -- Executing [s@macro-outisbusy:5] Congestion(SIP/306-00b8, 20)
 in new stack
 [2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862 ast_prod:
 Prodding channel 'SIP/306-00b8' failed
   == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
 'SIP/306-00b8' in macro 'outisbusy'
   == Spawn extension (from-internal, 0149XX, 7) exited non-zero on
 'SIP/306-00b8'
 -- Executing [h@from-internal:1] Hangup(SIP/306-00b8, ) in new
 stack
   == Spawn extension (from-internal, h, 1) exited non-zero on
 'SIP/306-00b8'
   == MixMonitor close filestream (mixed)
   == End MixMonitor Recording SIP/306-00b8


The verbose output states why your call is congested:

-- Got SIP response 556 No address found back from 217.195.XX.XXX:5060

The far end came back with a 556 response to the outbound INVITE
request. It doesn't think that whatever you dialled exists.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] how asterisk detects silence?

[Matthew Jordan]

On Mon, Mar 23, 2015 at 12:25 AM, Dmitry Melekhov d...@belkam.com wrote:
 19.03.2015 09:31, Dmitry Melekhov пишет:

 Hello!

 As I see there is  dsp_drop_silence switch in confbridge.
 Could you tell me how asterisk detects silence?
 Is it possible to change silence level,
 so, let's say some not loud enough background noises will be recognized as
 silence
 and only loud enough human voice will be recognized as sound?

 Thank you!


Asterisk passes received voice data, converted to signed linear,
through a DSP. The DSP looks at the energy level in the data and, if
it is above a certain value, categorizes the data as 'silence' or
'talking'.

You can tweak the periods necessary for Asterisk to decide if someone
is talking or silent using the 'dsp_silence_threshold' and
'dsp_talking_threshold' settings:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] PJSIP - Video Support for WebRTC

[Matthew Jordan]

On Mon, Mar 23, 2015 at 8:55 AM, Gosmac gosee...@gmail.com wrote:
 Hey i have an interesting topic to discuss here.

 The main goal here is to be able to make a video call between two WebRTC 
 endpoints registered on asterisk 13 it is a feature that definitely asterisk 
 13 should support .

 the problems that i faced with this is the following and i hope i could get 
 an advise here.

 asterisk 13 vanilla version has some issues marking the video packets this 
 complain web browser specially VP8 codecs so a friend of mine help me to 
 patch res_rtp_asterisk and now asterisk is marking video streams :) it just 
 mark video packets not touch anything else and web browser show video on web 
 page now I’m using online demo http://tryit.jssip.net/ is stable and get more 
 updates than sipml5. so i try echo() dialplan test and everything work 
 perfect on echo test :).

 i have two questions and i hope you could give me some advise.

 1) after marking video packet I’m able to make Dial() between two webrtc 
 peers but i get one way audio and video on callee party, “after 3 minutes on 
 call” i get two way audio and video on all parties seems to be not just a 
 problem on a missing keyframe.

  1.1) the 3 minutes delay only happen using chrome stable , could be a dtls 
 problem when asterisk make an offer to other endpoint?
  1.2) when i use chrome-dev and i disable dlts encryption everything work 
 perfect on video call.

 2) after marking video packets i realize that when you make a call with video 
 and you involve on dialplan an application like playback or music on hold any 
 application that  played audio files (audio and video never work).

 2.1) asterisk is muggling the audio and video streams ?

 This is good information for all guys out there that wants to support video 
 on webrtc in asterisk 13


Please stop spamming the list with this e-mail. Resending it multiple
times is clearly not yielding the results you'd like.


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Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

[Matthew Jordan]

On Thu, Mar 19, 2015 at 1:47 AM, Nick Awesome jl...@me.com wrote:
 NAT endpoint calling local endpount - switching to native_rtp then no audio,
 both of them have direct_media=no, Verbose log:

 -- Executing [99@dialmap:1] AGI(PJSIP/304-0022, /pbx/agi.php) in
 new stack
 -- Launched AGI Script /pbx/agi.php
 -- AGI Script Executing Application: (Dial) Options:
 (PJSIP/99/sip:99@192.168.1.73:5060,20)
 -- Called PJSIP/99/sip:99@192.168.1.73:5060
 -- PJSIP/99-0023 is ringing
 -- PJSIP/99-0023 answered PJSIP/304-0022
 -- Channel PJSIP/304-0022 joined 'simple_bridge' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
 -- Channel PJSIP/99-0023 joined 'simple_bridge' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
 Bridge da8840bc-9b71-4ca6-b1d8-9565bf8e5e28: switching from
 simple_bridge technology to native_rtp
 Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
 stack
 Locally RTP bridged 'PJSIP/99-0023' and 'PJSIP/304-0022' in
 stack
 0x7f4b50145420 -- Probation passed - setting RTP source address to
 194.204.157.200:8972
 0x7f4b5014f140 -- Probation passed - setting RTP source address to
 192.168.1.73:5004
 -- Channel PJSIP/304-0022 left 'native_rtp' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
 -- Channel PJSIP/99-0023 left 'native_rtp' basic-bridge
 da8840bc-9b71-4ca6-b1d8-9565bf8e5e28
 -- PJSIP/304-0022AGI Script /pbx/agi.php completed, returning 4


Correct - and per the log, they shouldn't be in a direct media bridge:

Locally RTP bridged 'PJSIP/99-0023' and
'PJSIP/304-0022' in stack
Locally RTP bridged 'PJSIP/99-0023' and
'PJSIP/304-0022' in stack

Locally RTP bridged means media is still flowing through Asterisk, it
just isn't being decoded and passed through the core.


-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk 13 : SILK codec ?

[Matthew Jordan]

On Thu, Mar 19, 2015 at 1:22 PM, Steve Murphy m...@parsetree.com wrote:

 On Wed, Oct 29, 2014 at 7:10 PM, sean darcy seandar...@gmail.com wrote:

 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


 I just checked at http://digium.com/en/products/asterisk/downloads,
 and of the Add-On Voice Codecs, g.729 is the only one available
 for Asterisk 13.

 The Siren 7/14 and SILK codecs offer only 12.X selections, and I proved
 today that the 12.x codecs will not load on Asterisk 13.

 a few weeks has turned into almost half a year now. Are these
 codecs no longer going to be available for 13 and up? Or, were they
 just overlooked in the day-to-day rush called life?


Yup, it's definitely taken a lot longer to get to this than I thought.
It is, however, still in the queue.

If you're curious about my thought process on this, it's pretty simple:
1) These codecs - while cool and useful - are not nearly as widely
used as say, codec_g729.
2) They require an upgrade to the new media improvements that were
made in 13, plus testing, etc. - which isn't hard but does take time.
3) And frankly, I'm putting Digium developer's time on improving open
source Asterisk, which means fixing more critical bugs. When there
aren't as many critical bugs, we can allocate someone's time to doing
the upgrade on those codecs.

If you'd like to make that picture change faster, you can always pitch
in on the issue tracker.

Matt

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

[Matthew Jordan]

On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome jl...@me.com wrote:
 Hey guys,

 have issues with reinvite, no matter what endpoint is calling asterisk
 always tries switch simple_bridge to native_rtp

  Bridge 0422bfa0-9d22-4bba-9108-a3f14d7d1cab: switching from simple_bridge
 technology to native_rtp

 in endpoints table “direct_media” sets to “no” on all endpoints but it
 doesn’t help.

 if native_rtp not work for some reason I have oneway audio. how can I fix
 this? if I add mix_monitor it works, but it’s not a right way to fix this
 issues.


A native_rtp bridge is used for more than direct media. It is also
used for local native bridging, that is, when you have two RTP capable
channels in a bridge and Asterisk does not require the media to flow
through its core. The bridge then just performs a packet to packet
swap between the two RTP capable channels.

Note that on verbosity 4, Asterisk will tell you if the bridge is
locally or remotely bridging the two channels.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] pjsip: outofcall_message_context

[Matthew Jordan]

On Wed, Mar 18, 2015 at 4:43 AM, Dmitriy Serov serov@gmail.com wrote:
 Hello.

 Is there an analog option outofcall_message_context for pjsip?
 or: how to determine that the call is an outbound text message?


The 'message_context' endpoint option [1] should provide what you're
looking for.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_message_context

-- 
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk switching bridge to native_rtp even with direct_media=no

[Matthew Jordan]

On Wed, Mar 18, 2015 at 9:53 AM, Nick Awesome jl...@me.com wrote:
 Well, it breaks audio for all NAT endpoints, how can I fix this?


Local (packet to packet) bridging should not do that. Remote (direct
media) can do that.

Can you confirm - by looking at a verbose level 4 log - how Asterisk
is bridging the two channels?

-- 
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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Matthew Jordan]

On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 I see that my asterisk is started with the -g option, the core file I cannot
 find on my system (find / -name core*)


I would suspect one of the following:

(1) Asterisk is not actually crashing.
(2) Something is deleting the core files.
(3) The core files are hiding really, really well.

Either way, if you can't get a backtrace, there isn't much we can do
to help with that problem.

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Re: [asterisk-users] Asterisk 13.2.0 Video issues

[Matthew Jordan]

On Mon, Mar 16, 2015 at 6:12 PM, Toufic Khreish (Gmail)
toufic.khre...@gmail.com wrote:
 Hello Matthew,

 I have compiled Asterisk 13.2 with the following compiler Flags enabled:
 DON'T_OPTIMIZE
 DEBUG THREADS
 BETTER_BACKTRACES


 My asterisk is running with the asterisk_script:
 root 24048 39.4  2.4 128564 50640 pts/1Sl   00:02   2:21
 /usr/sbin/asterisk -f -vvvg -c

 core show locks

 ===
 === 13.2.0
 === Currently Held Locks
 ===
 ===
 === pending lock# (file): lock type line num function lock
 name lock addr (times locked)
 ===
 ===

 When my asterisk crashes there is no file called core.

 The results of  gdb -se asterisk -ex bt full -ex thread apply all bt
 --batch -c core  /tmp/backtrace.txt

 /usr/src/asterisk-13.2.0/core: No such file or directory.
 No stack.

 What could be the problem ?


(1) Asterisk only generates a core file if started with the '-g' option

(2) Your core file may not be located in the directory that you are
running gdb from. You will need to find where the core file was
located - this is typically determined by
/proc/sys/kernel/core_pattern

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org

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