Re: [asterisk-users] asterisk-users Confbridge

[Matthew Fredrickson]

Sorry about the trouble.  Unsubscribed that user from the mailing lists.

Matthew Fredrickson

On Fri, Aug 7, 2020 at 9:20 PM Elizabeth  wrote:
>
> I'm online on this site!
> So contact me in my profile:
> here
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Re: [asterisk-users] Stir Shaken

[Matthew Fredrickson]

On Mon, Jul 13, 2020 at 2:34 PM Saint Michael  wrote:
>>
>> There is a big confusion here about Stir Shaken. It is NOT a provider issue. 
>> Un fact, all providers are whasing their hands and modifying their swihtches 
>> to pass-through the Signature. They cannot sign the call because then the 
>> become the responsible party for the call before the FCC, and liable for any 
>> illegal call. Every owner of a PBX that sends calls to the network, except 
>> if you use a trunk for the likes of Vonage, needs to sign their calls. So if 
>> you send calls with any kind of dialer and use DIDs, real or "borrowed", you 
>> need to get the signature service urgently or your business will stop 
>> terminating calls. You cannot self-sign, you cannot get around it, the calls 
>> will either go to straight to voicemail or fail. Even worse, the carries wil 
>> play a fake voicemail and charge you a fee, something that some already a 
>> are doing when they detect robocallig.
>
> Don't even think about Transnexus, because they use 302 Redirect with a  
> header, and no version of Asterisk supports it.  I am the only game in the 
> world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is 
> literally true. If you need to sign your calls to get through, with Asterisk, 
> you need to connect to my service. I am an approved Service Provider from the 
> FCC. If you keep thinking this is not happening, it is, and your business 
> will disappear overnight.
> The issue is that Vicidial, for example, does not provide res_odbc and 
> func_odbc, so you need to solve that first with Vicidial. Then you can apply 
> the code I provided earlier and your calls with have a legal, binding 
> signature. The carriers verify each signature and discard the ones that fail 
> the cryptography test.

Sounds like you're trying to sell/direct people towards a service that
you've created.  Feel free to do so on the -biz list but the -users
list isn't the right place for that sort of thing.

Matthew Fredirckson

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Re: [asterisk-users] Stir Shaken is upon us

[Matthew Fredrickson]

On Sun, Jul 12, 2020 at 5:18 PM Joshua C. Colp  wrote:
>
> On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins  wrote:
>>
>> Asterisk 18 will have support based on this asterisk update Matt F did for 
>> CommCon's sponsor slots
>>
>> https://youtu.be/eas1csaX-wc
>>
>
> As well support will go into Asterisk 16 and 17 as well. It's just been under 
> active development so we've been waiting for that to finish before bringing 
> it back into those versions.
>

Thanks for clarifying that Josh.  I only had 5 min on the CommCon
presentation so I focused more on the Asterisk 18 side of things
rather than clarifying a lot of that :-)

Matthew Fredrickson

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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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[asterisk-users] Asterisk Usage Survey

[Matthew Fredrickson]

Hey All,

For those of you that do not know me, my name is Matthew Fredrickson
and I’m the project lead for the Asterisk project. First off, I wanted
to thank all of you that contribute in various ways to the project –
whether it be at a developmental level, answering questions on forums
and mailing lists, contributing documentation, or just generally
advocating for it within your sphere of influence. It takes so many
people’s efforts to make the project what it is and to sustain such a
large and vibrant user and developer community.

We created a general survey inquiring how people utilize Asterisk. It
should only take about 10-15 minutes, but would help us understand
better how our users are utilizing Asterisk and help us to understand
if there are important areas of Asterisk that we underemphasize from a
development perspective. If you don’t mind filling it out, it would be
greatly appreciated.

Thanks *so* much again for your time, and best wishes to each of you
in your efforts.

https://goo.gl/forms/xL1VUHRsf95saly13

Matthew Fredrickson

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Re: [asterisk-users] tel URI

[Matthew Fredrickson]

On Thu, Jan 31, 2019, 9:24 AM Jean-Denis Girard  Hi list,
>
> Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a
> system that uses exclusively tel: uri on inbound and outbound calls. I
> could not find documentation or sample config about tel:uri. Is this
> doable? If not possible with PJSIP, is chan_sip a better option? Any
> pointer would be greatly appreciated.
>

Right now, chan_pjsip does not properly handle tel: URIs. If you need them
you might need to use chan_sip.

Matthew Fredrickson


>
> Thanks,
> --
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>
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Re: [asterisk-users] Stop

[Matthew Fredrickson]

Unsubscribe info is in the footer of the message 😀

Best wishes,
Matthew Fredrickson

On Mon, Oct 1, 2018, 6:22 AM Karen York  wrote:

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Re: [asterisk-users] WebRTC as Softphone substitute ?

[Matthew Fredrickson]

On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez  wrote:
>
> On 9/26/2018 4:46 AM, Olivier wrote:
>
> > Hello,
> >
> > This morning, I asked myself if WebRTC could be a viable alternative
> > to softphone deployment.
> >
> > For me, main issue with Softphones is the amount of work needed for
> > installation and configuration.
> > Also, Softphones must be carefully choosen if Deskphone-like quality
> > is expected.
> >
> > Now that WebRTC becomes ubiquitous, it might make sense to trade
> > Softphone features (call history, BLF, ...) for WebRTC deployment
> > simplicity.
> >
> > What do you think of this ?
> > What kind of experience did you met with such WebRTC deployments ?
> > What about classic telephony features (CallTransfer) ?
> > Have you tried Cyber Maga Phone 2K ?
> >
>
>  If you can get it to work WebRTC is a good option.  The problem is
> that any changes in your network may disrupt it and even trying to
> replicate your installation is difficult.  I have it working fine on my
> website so customers can call us directly from our web page but I never
> could get Cyber Mega Phone 2K to work on the same server.  We used JSSIP
> to create the webrtc phone on our website.

We just updated the documentation for how to get CMP2K working on the
wiki [1].  We'd love some feedback if you still have issues getting it
setup so that we can improve the docs.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone

Best wishes,
Matthew Fredrickson

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Re: [asterisk-users] AGI timeout option

[Matthew Fredrickson]

Hey,

I would suggest starting a new thread with this question instead of
inserting this into another existing thread like this.

Matthew Fredrickson

On Tue, Sep 18, 2018, 11:16 AM modou lo  wrote:

> Please can i ask you i want to know which code can help me to provide the
> taxation of voip/toip services in asterisk
>
> Le mar. 18 sept. 2018 à 01:36, Patrick Wakano  a
> écrit :
>
>> Thanks everyone for the answers!
>> I did explored some options at the PHP level and probably will do
>> something in this direction, but in fact what I was really looking was
>> something in the Asterisk side, not in the script side.
>> Because in my opinion regardless of the language or AGI type, Asterisk
>> itself should be able to timeout a long running script and return to the
>> dialplan. However looks like there is nothing of this sort.
>>
>> Kind regards,
>> Patrick Wakano
>>
>> On Sat, 15 Sep 2018 at 03:56, Eric Wieling  wrote:
>>
>>> I don't know AGIspeedy, but I have some PHP scripts where I set a
>>> connect timeout using streams.
>>>
>>> Example using https, but should be easily adaptable to non-s http.:
>>>
>>> $pbxsh_bin = @file_get_contents("https://blah.blah.blah";, FALSE,
>>> @stream_context_create(array('https' => array('timeout' => 5,
>>> "verify_peer"=>false, "verify_peer_name"=>false;
>>>
>>> On 09/14/2018 01:40 PM, Carlos Chavez wrote:
>>> > On 9/13/2018 8:04 PM, Patrick Wakano wrote:
>>> >
>>> >> Hello list,
>>> >> Hope you all doing  well!
>>> >>
>>> >> Recently, I had an issue with a FastAGI PHP script, which under some
>>> >> specific situation would run into an infinity loop, consuming all CPU
>>> >> resources. This also was preventing Asterisk to terminated the call
>>> >> properly because it was waiting for the AGI to return... The
>>> >> application uses AGIspeedy to process the AGI calls, not sure if this
>>> >> can be affecting this situation somehow
>>> >> Due to this problem I started looking for some option to timeout the
>>> >> AGI call and return to the dialplan after XYZ seconds and so this
>>> >> would protect Asterisk preventing the dialplan to get stuck due to
>>> >> some external script problem that is actually outside of Asterisk
>>> >> control. Does Asterisk provide some control of this sort? I searched
>>> >> around and could not find any.
>>> >> Any idea is appreciated!
>>> >>
>>> >> Kind regards
>>> >> Patrick Wakano
>>> >>
>>> >
>>> > I think this is what you may be looking for:
>>> >
>>> > http://php.net/manual/en/function.set-time-limit.php
>>> >
>>>
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>>>
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Re: [asterisk-users] Asterisk 16 AMI changes

[Matthew Fredrickson]

Usually yes. You'll need to read the UPGRADE.txt and CHANGES files to get a
good idea of the specific changes though.

Best wishes,
Matthew Fredrickson

On Thu, Sep 6, 2018, 7:44 PM Telium Support Group  wrote:

> Does anyone know if Asterisk 16 includes changes to the AMI?  (syntax /
> commands / etc)
>
>
>
> I see a release candidate is forthcoming.  Just curious
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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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> >
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> >
> >  New to Asterisk? Start here:
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> >
> >  asterisk-users mailing list
> >  To UNSUBSCRIBE or update options visit:
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> > [2  ]
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> >
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> >
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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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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
&

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

[Matthew Jordan]

 is available at:
> > https://www.asterisk.org/community/astricon-user-conference
> >
> > Check out the new Asterisk community forum at:
> > https://community.asterisk.org/
> >
>
> I agree. That's why I hacked chan_sip.c to get the addresses in the log.
>
> I'm surprised they're not in the log by default. I must be the only person
> who gets these "non-critical invites".
>
> sean
>
>
>
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> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>
>
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>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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-- 
Matthew Jordan
Digium, Inc. | CTO
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-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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> asterisk-app-...@lists.digium.com
> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev
>


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Digium, Inc. | CTO
445 Jan Davis Drive NW - Hunts

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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>> asterisk-app-...@lists.digium.com <mailto:asterisk-app-...@lists.digium.com>
>> http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app

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 
> 
> ___
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Re: [asterisk-users] More testing

[Matthew Fredrickson]

:-) Sorry to disappoint.

On Wed, May 23, 2018, 10:21 AM John Kiniston  wrote:

> I got excited when I saw 8 new messages on the Asterisk list-serve this
> morning, What discussions must be happening I thought!
>
> You are a tease sir.
>
> On Tue, May 22, 2018 at 7:58 PM, Matt Fredrickson 
> wrote:
>
>> More testing.  Test test test. :-)
>>
>> --
>> Matthew Fredrickson
>> Digium, Inc. | Engineering Manager
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> 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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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Testing for real from a non-digium email

[Matthew Fredrickson]

Here we go!

Matt

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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  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.

-- 
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] PJSIP_AOR Slow

[Matthew Jordan]

On Sun, Nov 19, 2017 at 5:38 AM, Daniel Journo  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]

gt; Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> 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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>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
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Re: [asterisk-users] RTCP + Stasis causing high memory consumption

[Matthew Jordan]

 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  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  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
 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  wrote:

>
>
> 2017-02-16 14:27 GMT+01:00 Joshua Colp :
>
>> 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  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 
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 
wrote:

> On 21-11-16 15:17, Matthew Jordan wrote:
>
>
> On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens 
> 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 
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  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  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 
wrote:

> On Thu, 10 Nov 2016 00:35:54 +0100
> Max Grobecker  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  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  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  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  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  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  wrote:
> Are you sending progress?
>
>
> בתאריך 24 באוג׳ 2016 13:40,‏ "Saint Michael"  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]

_device_state: No provider found,
> checking channel drivers for SIP - 100
> 30 DEBUG[-1]: chan_sip.c: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  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  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  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  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  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  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  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  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  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 
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] SIP/SDP for MulticastRTP page

[Matthew Murphy]

Thanks Josh,

I have actually built my own endpoints and was experimenting with dynamically 
creating multicast sessions so that I didn't need to pre-configure the 
multicast addresses at all. When you say, "...This eliminates the need to set 
up a SIP session for each device to have them listen in, which can be 
problematic." What do you mean by "problematic"? I was just curious. I thought 
SDP was built for this kind of thing, but I don't know the history and I am 
sure there are things I haven't thought of when it comes to implementation, 
security, etc.

Also, do you have any thoughts on setting up multicast sessions without a 
priori knowledge on the endpoints? Would I have to spin my own message protocol 
to do this? Could I monkey around in the Asterisk source to make it work? Or, 
is it just a huge waste of time and effort?

I really appreciate your quick response to my earlier question. Thanks a lot 
for your time.

--Matt


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Joshua Colp 

Sent: Wednesday, April 27, 2016 9:58:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP/SDP for MulticastRTP page

Matthew Murphy wrote:
> Hi everyone,
>
>
> I am sending out a multicast page using the following in my dialplan:
>
>
> Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)
>
>
> Everything works great, but I had a question about SIP and SDP:
>
>
> Should I be seeing a SIP/SDP message from the asterisk server containing
> media information and the multicast IP address? On wireshark, I see
> SIP/SDP from the admin phone I am using to dial the extension and
> initiate the page. But I never see a SIP/SDP message with the multicast
> address sent from the Asterisk server to the endpoints. Maybe I
> misunderstand how SIP and SDP fit into the messaging scheme.

You won't. It's up to the phones to be configured to always listen to
the multicast address and play it out over the speakerphone. This
eliminates the need to set up a SIP session for each device to have them
listen in, which can be problematic.

Cheers,

--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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[asterisk-users] SIP/SDP for MulticastRTP page

[Matthew Murphy]

Hi everyone,


I am sending out a multicast page using the following in my dialplan:


Page(MulticastRTP/linksys/${MULTICAST_IP_ADDR}:6061/${MULTICAST_IP_ADDR}:6061,q)


Everything works great, but I had a question about SIP and SDP:


Should I be seeing a SIP/SDP message from the asterisk server containing media 
information and the multicast IP address? On wireshark, I see SIP/SDP from the 
admin phone I am using to dial the extension and initiate the page. But I never 
see a SIP/SDP message with the multicast address sent from the Asterisk server 
to the endpoints. Maybe I misunderstand how SIP and SDP fit into the messaging 
scheme.


Can anyone tell me if I should see SIP/SDP coming from my Asterisk server to my 
endpoints? I hope my question makes sense.


Thanks,


--Matt
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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 
wrote:

> Hello,
>
> I'm using the following Dial command syntax:
> Dial*(SIP/peer/exten!sip:x...@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 *, 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 
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
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 13.6.0: 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 
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" 
> *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
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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[asterisk-users] Asterisk 13.7.0 losing database connection

[Matthew Murphy]

Hi everyone,


I upgraded from Asterisk 13.5.0 to 13.7.0 and I am having database connection 
problems. I am doing Asterisk realtime with PJSIP 2.4.5 and it works perfectly 
in 13.5.0. But now I am losing my database connection (running on a virtual 
box) and I am stuck!


I spent all day yesterday looking through information from people that had 
similar problems. So far as I can tell, this pops up from time to time, but I 
never saw any resolution. I've tried to increase connection_timeout and monkey 
around with bind-address but no joy.


Has anyone had any experience with this?



Below is a snippet from my CLI that shows the problem:



[Jan 26 09:58:27] WARNING[23947]: res_odbc.c:661 ast_odbc_prepare_and_execute: 
SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]...

[Jan 26 09:58:27] WARNING[23947]: res_odbc.c:765 ast_odbc_sanity_check: 
Connection is down attempting to reconnect...

Error in my_thread_global_end(): 1 threads didn't exit

[Jan 26 09:58:32] NOTICE[23947]: res_odbc.c:1528 odbc_obj_connect: Connecting 
asterisk

[Jan 26 09:58:32] NOTICE[23947]: res_odbc.c:1567 odbc_obj_connect: res_odbc: 
Connected to asterisk [asterisk-connector]

[Jan 26 09:58:32] WARNING[23947]: res_odbc.c:649 ast_odbc_prepare_and_execute: 
SQL Execute returned an error -1: 23000: [MySQL][ODBC 5.1 
Driver][mysqld-5.5.46-0+deb8u1]Duplicate entry 
'103^3B@b2ddb71120ca050b70435cd8679200ba' for key 'id' (118)

[Jan 26 09:58:32] WARNING[23947]: res_odbc.c:661 ast_odbc_prepare_and_execute: 
SQL Execute error -1! Verifying connection to asterisk [asterisk-connector]...

[Jan 26 09:58:32] WARNING[23947]: res_odbc.c:765 ast_odbc_sanity_check: 
Connection is down attempting to reconnect...

Error in my_thread_global_end(): 1 threads didn't exit


Thanks,


--Matt
-- 
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Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)

[Matthew Murphy]

I compiled Asterisk 13.5.0 (and 13.7.0-rc2), PJproject 2.4.5, and DAHDI 2.11.0 
from source. I am using Debian 8.2 and mysql is version 5.5.46.

This is a recent problem and when we went back to 13.5.0 and started seeing the 
issue, we started thinking that mysql must have updated and caused the problem. 
I think you are spot-on with your analysis.

I will do a little experimentation with mysql and see if I can isolate where 
the problem started.

Thanks,

--Matt

From: asterisk-users-boun...@lists.digium.com 
 on behalf of A J Stiles 

Sent: Tuesday, January 19, 2016 7:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2
(libmysqlclient?)

On Monday 18 Jan 2016, Matthew Murphy wrote:
> Hi everyone,
>
> I am getting a segmentation fault (seems to occur randomly) using Asterisk
> 13.7.0-rc2 with PJProject 2.4.5. It appears to be something that
> libmysqlclient is complaining about when doing a query in
> ps_endpoint_id_ips. We are using Asterisk Realtime. This also seems to
> occur in Asterisk 13.5.0.

Which bits did you compile from Source Code yourself, and which bits  (if any)
are precompiled by your distribution?  What is your libmysqlclient version?

If you are using some sort of Ubuntu / Debian-based distribution, could it
possibly have done a sneaky `apt-get upgrade` behind your back?

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

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[asterisk-users] Segmentation Fault Asterisk 13.7.0-rc2 (libmysqlclient?)

[Matthew Murphy]

Hi everyone,

I am getting a segmentation fault (seems to occur randomly) using Asterisk 
13.7.0-rc2 with PJProject 2.4.5. It appears to be something that libmysqlclient 
is complaining about when doing a query in ps_endpoint_id_ips. We are using 
Asterisk Realtime. This also seems to occur in Asterisk 13.5.0.


Below is a backtrace that might help a little. I have looked through the change 
log for the 13.7.0 release and some of items addressed may fix my problem. 
Before diving in and attempting to upgrade to the final version of 13.7.0, I 
was hoping someone with knowledge would be able to look at this and let me know 
if this is something already seen or if this is entirely new.


Thanks for the help!


--Matt


---
BACKTRACE BELOW
---




[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `asterisk -g'.
Program terminated with signal SIGSEGV, Segmentation fault.
#0  0x7f1e02e8a120 in list_add () from 
/usr/lib/x86_64-linux-gnu/libmysqlclient.so.18
#0  0x7f1e02e8a120 in list_add () from 
/usr/lib/x86_64-linux-gnu/libmysqlclient.so.18
No symbol table info available.
#1  0x7f1e0339d132 in my_SQLAllocStmt () from 
/usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so
No symbol table info available.
#2  0x7f1e38354af4 in ?? () from /usr/lib/x86_64-linux-gnu/libodbc.so.2
No symbol table info available.
#3  0x7f1e01e8f16f in custom_prepare (obj=0x1e588c8, data=0x7f1dc8a289b0) 
at res_config_odbc.c:107
res = 0
x = 1
count = 0
cps = 0x7f1dc8a289b0
field = 0x1e588a8
encodebuf = 
"\001\200\255\373\000\000\000\001P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\206\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000P\205\242\310\035\177\000\000\237\205\242\310\035\177\000\000\342X\350\002\036\177\000\000\247p\350\001\000\000\000\000\351ʎ:\036\177\000\000\000\000\000\000\000\000\000\000\335Y\350\002\036\177\000\000`\000\000\000\004\000\000\000\a\000\000\000\000\000\000\000@\256\350\001\000\000\000\000\a",
 '\000' , 
"\340U\347\002\036\177\000\000\a\000\000\000\000\000\000\000;\000\000\000\000\000\000\000\002\000\000\000\035\177\000\001`\204\242\310\035\177\000\000\377\377\377\377\000\000\000\000\000"...
stmt = 0x0
__PRETTY_FUNCTION__ = "custom_prepare"
#4  0x7f1e385b9783 in ast_odbc_prepare_and_execute (obj=0x1e588c8, 
prepare_cb=0x7f1e01e8f11f , data=0x7f1dc8a289b0) at 
res_odbc.c:640
res = 0
i = 0
attempt = 0
nativeerror = 0
numfields = 0
diagbytes = 0
state = "\v:\351\001\036\177\000\000t\001"
diagnostic = 
"\340\030\006\244\035\177\000\000̜\242\310\035\177\000\000\200\020\024\002\000\000\000\000,Z_9\036\177\000\000)\266\227V\000\000\000\000\320\003\000\000\000\000\000\000\240\210\242\310\035\177\000\000\350\341\000D\035\177\000\000\300s\276\314\035\177\000\000\255I^\000\000\000\000\000\350\341\000D\035\177\000\000p>d\000\000\000\000\000̜\242\310\034\b\000\000K6d\000\000\000\000\000\320\003\000\000\000\000\000\000\001\000\000\000\000\000\000\000`\211\242\310\035\177\000\000Г\002\244\035\177\000\000\360\210\242\310\035\177\000\000\065\200^\000\000\000\000\000@F\351\001\036\177\000\000`\211\242\310t\001\000\000\v:\351\001\036\177\000\000\000\001\000\000\000\000\000\000\300\211\242\310\035\177\000\000ȑ\242\310"...
stmt = 0x7f1dc8a297b0
__PRETTY_FUNCTION__ = "ast_odbc_prepare_and_execute"
#5  0x7f1e01e90715 in realtime_multi_odbc (database=0x7f1dc8a298d0 
"asterisk", table=0x7f1dc8a297d0 "ps_endpoint_id_ips", fields=0x7f1da40185f0) 
at res_config_odbc.c:376
obj = 0x1e588c8
stmt = 0x25ed6b2
sql = "SELECT * FROM ps_endpoint_id_ips WHERE id LIKE ? ORDER BY 
id\000\177\000\000\377\377\377\377\377\377\377\377\000\000\000\000\000\000\000\000`\224\242\310\035\177\000\000m\225\242\310\035\177\000\000`\225\242\310\035\177\000\000`\226\242\310\035\177\000\000@\225\242\310\035\177\000\000\033\311d9\036\177\000\000P\225\242\310\035\177\000\000ȕ\242\310",
 '\000' , "\377\377\377\377", '\000' , " 
R\234\317\035\177\000\000\000\000\000\000\035\177", '\000' ...
coltitle = "\002", '\000' , 
"|\000\000\000\000\000\000\000ǫ\b\316", '\000' , "\002", 
'\000' , 
"\021\000\000\000\000\000\000\000\035\177\000\000\000\000\000\000\000\000\000\000
 
\000\000\000\000\062\062\062\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\035\177\000\000\000\000\000\000\036\177\000\000\377\377\377\377\035\177\000\000\000\000\000\000\036\177",
 '\000' , "\377\377\377\377\377\377\377\377%", '\000' 
, 
"\372I\\9\036\177\000\000\000\224\242\310\035\000\000\000\062"...
rowdata = 0x7f1da4001110
initfield = 0x7f1dc8a28960 "id"
op = 0x7f1e01e93b2d ""
field = 0x0
stri

Re: [asterisk-users] PJSIP Returning 421 Extension Required

[Matthew Jordan]

On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard  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: 
> f: ;tag=10847511385389740959
> i: 117620342110831512016142@10.77.27.103
> CSeq: 1 INVITE
> d: no-fork
> Privacy: none
> P-Asserted-Identity:
> 
> 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:  ;user=phone>;tag=10847511385389740959
> To:  ;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] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2

[Matthew Murphy]

I am using realtime - you got it! 

Great to know it already has a fix. I'll pull in rc3 and go from there.

Thanks a lot for your help!

--Matt


From: asterisk-users-boun...@lists.digium.com 
 on behalf of Joshua Colp 

Sent: Wednesday, January 13, 2016 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] "pjsip show endpoints" returns "No Objects Found" 
in 13.7.0-rc2

Matthew Murphy wrote:
> Hi everyone,
>
>
> I have just upgraded to *Asterisk 13.7.0-rc2* and noticed that when I
> type "/pjsip show endpoints/" at the CLI, I get "/No Objects Found/".

Are you using realtime? A regression was found[1] in rc2 when realtime
was in use which would cause this to happen. Only contacts were
mentioned but it would impact other things. There is now an rc3 which
has a fix for this in it.

[1] https://issues.asterisk.org/jira/browse/ASTERISK-25689

--
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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[asterisk-users] "pjsip show endpoints" returns "No Objects Found" in 13.7.0-rc2

[Matthew Murphy]

Hi everyone,


I have just upgraded to Asterisk 13.7.0-rc2 and noticed that when I type "pjsip 
show endpoints" at the CLI, I get "No Objects Found".


However, if I request information on a specific endpoint, (for example: "pjsip 
show endpoint 101") then I get all of the information for that endpoint as 
expected.


This seems to have started as soon as I upgraded to 13.7.0-rc2. I tried with 
pjproject 2.4 and now pjproject 2.4.5 and get the same result.


Has anyone else seen this or is it something that is unique to my situation?


Thanks,


--Matt
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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  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
>

Re: [asterisk-users] Asterisk CLI and database problem

[Matthew Jordan]

On Tue, Dec 22, 2015 at 3:48 AM,  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


-- 
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Re: [asterisk-users] DIALSTATUS not being set

[Matthew Jordan]

On Tue, Dec 22, 2015 at 7:26 AM, Marcos Prates 
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-'
-- 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] CEL entries over ODBC several hours late (Matthew Jordan)

[Matthew Jordan]

On Thu, Dec 10, 2015 at 8:57 AM, Stefan Viljoen 
wrote:

> Hi Matthew
>
> Thank you very much for the reply.
>
> I must have something seriously wrong somewhere else then - I retested now
> and the "apparent" effect is as  I describe but your info definitely
> contradicts that. But you're obviously correct.
>
> One more question - I've noted that if I run a combination of queries in
> the
> CEL backing DB (MariaDB) and the CEL table is locked, this severely affects
> the Asterisk instance - thousands of occurrences of
>
> chan_sip.c:4057 __sip_autodestruct: Autodestruct on dialog
> '6a9f5d3543b619655e07c81437373a32@172.17.12.3:5060' with owner
> SIP/3034-000207c8 in place (Method: BYE). Rescheduling destruction for
> 1
> ms
>
> appear in the CLI and users complain that if they hang up then they cannot
> make another call on the same SIP handset for several minutes.
>
> This is obviously because the dialplan gets delayed in the H extension, and
> cannot write to the CEL table, waiting for the MariaDB instance to clear
> the
> locks so it can write again. The above apparently comes from a watchdog
> process that watches how "fast" the H extension is and if it takes "too
> long" it forces the channels closed.
>
> Is this assumption correct?
>
> Addtionally, it seems that the writing of CEL in Asterisk is NOT async?
> E.g.
> it appears the thread that was running the conversation ALSO does the CEL
> writing / pushing to the CEL core as you describe in a synchronous manner.
>
> For 1.8, is this correct that CELs are synchronous, and do newer Asterisk
> versions do it async?
>
>
That's actually a bit surprising. Assuming you are using CELGenUserEvent in
the 'h' extension, I would not expect that to block the channel when
writing out to the database. The act of creating the CEL event will lock
the channel briefly, but the actual CEL event is queued up onto a message
bus and dispatched to another thread. I'd have to see the output of a gdb
backtrace or 'core show locks' to know why that is impacting the channels.

It should be asynchronous everywhere - in 1.8+. While the implementation of
the message bus changed in Asterisk 12, that doesn't change the nature of
how it is dispatched. As I said, a gdb backtrace or 'core show locks' would
show who the culprit it.


E. g. my point being if there are major DB issues, it is quite a bit
> kryptonite to have that "spill back" into Asterisk and start blocking off
> users from calling out - wouldn't it be much better to simply have failed
> CEL writes just die in a distal thread instead of the main call thread for
> the channels running on that handset.
>
> Or am I completely misunderstanding things?
>
> Anyway, thanks for the reply. :)
>
> Kind regards,
>
> > 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 aste

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

[Matthew Jordan]

On Wed, Dec 9, 2015 at 6:32 AM, Stefan Viljoen 
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

-- 
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] host parameter equivalent in pjsip.conf

[Matthew Jordan]

On Tue, Dec 8, 2015 at 10:29 AM, xaled  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: ;tag=as5152122a
>
> To: 
>
> Contact: 
>
> 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: ;tag=f47f3ed2-0975-4ff0-bd3b-bd5c38e594c4
>
> To: 
>
> Contact: 
>
> 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

-- 
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] after upgrade buttons on Dahdi phones don't work [SOLVED]

[Matthew Jordan]

On Sat, Dec 5, 2015 at 1:17 PM, Greg Woods  wrote:

>
>
> On Fri, Dec 4, 2015 at 12:50 PM, Greg Woods  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/

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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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 
wrote:

> On Mon, 30 Nov 2015 09:40:50 -0600
> Matthew Jordan  wrote:
>
> > On Sat, Nov 28, 2015 at 7:14 AM, Ethy H. Brito 
> > 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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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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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 
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

-- 
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 and RTT in realtime

[Matthew Jordan]

On Thu, Oct 29, 2015 at 2:34 PM, Ryan, Travis  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
 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  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] Help with voicemail

[Matthew Jordan]

On Sat, Oct 17, 2015 at 10:12 AM, Luca Bertoncello  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] Sending XML over the asterisk PJSIP

[Matthew Jordan]

On Sat, Oct 17, 2015 at 2:32 AM, Thyda ENG  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  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

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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  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] Semicolon use in configuration?

[Matthew Jordan]

On Sun, Oct 11, 2015 at 8:55 PM, Juan van Rooyen
 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  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  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] Change Asterisk MulticastRTP codec

[Matthew Murphy]

Larry and Pete,
Thanks a bunch for jumping in and giving me some ideas! I am hoping to have 
something working soon with what you guys have given me. The end game for me is 
to be able to stream MP3s from a playlist. It appears like both solutions you 
guys have proposed may give me what I need. I will actually try both and let 
you know how it goes.
 --Matt

From: lmo...@omninet.net.au
To: asterisk-users@lists.digium.com
Date: Thu, 1 Oct 2015 06:15:17 +0800
Subject: Re: [asterisk-users] Change Asterisk MulticastRTP codec


  

  
  
On my Asterisk 11 system I have the following in extensions.ael for
chan_sip.



8001=> {

Set(SIP_CODEC=alaw);

   
//Dial(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061);

   
Page(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061,q,5);

Hangup();

};





I believe in your case you need to set PJSIP_MEDIA_OFFER(ulaw) in a
pre-dial handler prior to making the call.



See
https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification.









On 1/10/2015 1:51 AM, Matthew Murphy
  wrote:



  
  Greetings everyone,



I was wondering if there was a way to change the codec that
  Asterisk uses when streaming via MulticastRTP. Or perhaps a
  way to transcode the multicast stream.



In the CLI, when I have a multicast stream in progress, I
  am typing 'core show channel MulticastRTP/0x7f7' to
  get lots of helpful information.



I have noticed that when I do

  a MULTICAST page and send data from MP3Player, I get no
  sound on my speakers and get the following from 'core show
  channel PJSIP/xxx':




  NativeFormats: (slin)
  WriteFormat: slin
  ReadFormat: slin
  WriteTranscode: No 
  ReadTranscode: No 




I have noticed that when I do a UNICAST page and send
data from MP3Player, everything works flawlessly and I
  get the following from 'core show channel MulticastRTP':




  NativeFormats: (ulaw)
  WriteFormat: slin
  ReadFormat: slin
  WriteTranscode: Yes (slin@8000)->(ulaw@8000)
  ReadTranscode: Yes (ulaw@8000)->(slin@8000)







The only thing that is changing is the following
  line in my extensions.conf file:



; For Multicast Paging

  same =>
n(next),Page(MulticastRTP/basic/239.1.1.2:6061,q)
  

  
  ; For Unicast Paging
  same =>
n(next),Page(PJSIP/107&PJSIP/108,ib(pjsip-auto-answer-header^addheader^1)|p})







Is there any way to get the MP3Player stream to transcode
  (as it does on the UNICAST stream) when I try to MULTICAST?



Thanks for the help,



--Matt
  
  

  
  




  


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[asterisk-users] Change Asterisk MulticastRTP codec

[Matthew Murphy]

Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when 
streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show 
channel MulticastRTP/0x7f7' to get lots of helpful information.
I have noticed that when I do a MULTICAST page and send data from MP3Player, I 
get no sound on my speakers and get the following from 'core show channel 
PJSIP/xxx':
NativeFormats: (slin)WriteFormat: slinReadFormat: slinWriteTranscode: No 
ReadTranscode: No 
I have noticed that when I do a UNICAST page and send data from MP3Player, 
everything works flawlessly and I get the following from 'core show channel 
MulticastRTP':
NativeFormats: (ulaw)WriteFormat: slinReadFormat: slinWriteTranscode: Yes 
(slin@8000)->(ulaw@8000)ReadTranscode: Yes (ulaw@8000)->(slin@8000)

The only thing that is changing is the following line in my extensions.conf 
file:
; For Multicast Pagingsame => n(next),Page(MulticastRTP/basic/239.1.1.2:6061,q)
; For Unicast Pagingsame => 
n(next),Page(PJSIP/107&PJSIP/108,ib(pjsip-auto-answer-header^addheader^1)|p})

Is there any way to get the MP3Player stream to transcode (as it does on the 
UNICAST stream) when I try to MULTICAST?
Thanks for the help,
--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  wrote:
> On Mon, 21 Sep 2015 06:48:52 +
> Emil Ohlsson  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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Digium, Inc. | Director of Technology
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] AMI 'meetme list concise' hanging

[Matthew Jordan]

On Mon, Aug 31, 2015 at 10:52 AM, Steve Edwards
 wrote:
> I have a problem with AMI 'meetme list concise' hanging. I'm running
> Asterisk 11.15.1, and PHPAGI 2.20.
>
> The AMI stuff is in the file phpagi-asmanager.php, which says it is v 1.10
> 2005/05/25.
>
> Here's the relevant snippet of my PHP code:
>
> // get list of conferences
> if  ($debug)
> {
> echo 'getting list of conferences' . PHP_EOL;
> }
> $response = $ami->command("meetme list concise");
> if  ($debug)
> {
> echo 'got list of conferences' . PHP_EOL;
> }
>
> I never get to 'got.'
>
> 1) Is this the current / best version of the PHP AMI interface?
>
> 2) Is this a 'known issue' with either Asterisk 11.15.1 or PHPAGI 2.20?

I'm not aware of this issue in a recent version of Asterisk (where
11.15.1 is relatively recent). It may be worth looking at the Asterisk
logs to see if the command is returning. If so, then the issue may
just be in the underlying PHP library.

> 3) Can I set some sort of timer to abort the AMI command if it takes longer
> that a second or two?
>
> 4) Is there a better 'work-around?'
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>
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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  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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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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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  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: 
>  From: ;tag=284759743 
> To:  Call-ID: 76603@192.168.1.100 CSeq: 542 
> MESSAGE Contact:  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: 
> ;tag=284759743 To: 
> ;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.

-- 
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] Changing volume via dialplan

[Matthew Murphy]

You are right, it works just fine!
Needless to say, I had tricked myself into thinking I was using Asterisk 13.5. 
There was a build error getting covered up by my install script. So I was still 
using the "old" Asterisk 13.4.0 that I had previously installed. Once I fixed 
that, everything worked nicely.
Typical PICNIC error: Problem In Chair, Not In Computer.
Thanks for the help!

> Date: Tue, 25 Aug 2015 15:00:09 -0300
> From: jc...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Changing volume via dialplan
> 
> Matthew Murphy wrote:
> > Greetings everyone,
> 
> Kia ora,
> 
> > I am attempting to adjust the volume of a call using *Set(VOLUME)* in my
> > extensions.conf file. I am finding that*Set(VOLUME(TX)=x)*and
> > *S**et(VOLUME(RX)=y)*have no discernable effect on my endpoints (Snom
> > 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0,
> > 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone
> > volume. I can see that the Set(VOLUME) instruction is being executed on
> > the Asterisk CLI.
> 
> I just did the following on 13.5.0:
> 
> exten => 1002,1,Answer
> exten => 1002,2,Set(VOLUME(tx)=10)
> exten => 1002,3,Playback(demo-congrats)
> 
>  From my D70 and confirmed the audio was ... louder/horrible.
> 
> Do you have any other endpoints you could test from?
> 
> >
> > I have also tried using *Set(CHANNEL(txgain)=x)* and
> > *Set(CHANNEL(rxgain)=y)* and those don't seem to have any effect either.
> > I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10,
> > and 100 and there appears to be no change on the phone volume. I can see
> > that the Set(CHANNEL) instruction is being executed on the Asterisk CLI.
> 
> These aren't applicable to PJSIP.
> 
> >
> >
> > I am using *PJSIP *and just upgraded to *Asterisk 13.5.0*. It wasn't
> > working on 13.4.0 either, but when I saw the release notes on 13.5 and
> > volume was addressed, I was hopeful that it might solve my problem for me.
> >
> >
> > So I have a couple of questions:
> >
> >
> > 1) Am I using the correct functions in the dial plan to adjust volume?
> > It would be something like:
> >
> > same => n,Set(VOLUME(TX)=3)
> >
> > or
> >
> > same =>n,Set(CHANNEL(rxgain)=0)
> 
> Yes, the VOLUME dialplan function should do the job.
> 
> >
> > 2) If this is correct, what are the min/max values that I can use when
> > adjusting volume? I was digging around the source code and it looked
> > like maybe a min of -4 and max of +4 was expected - but I am unsure.
> 
> There is no enforced minimum/maximum. The value provided is in dB.
> 
> Cheers,
> 
> -- 
> 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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[asterisk-users] Changing volume via dialplan

[Matthew Murphy]

Greetings everyone,
I am attempting to adjust the volume of a call using Set(VOLUME) in my 
extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) 
have no discernable effect on my endpoints (Snom 300 IP phones). I have tried 
setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there 
appears to be no change on the phone volume. I can see that the Set(VOLUME) 
instruction is being executed on the Asterisk CLI.
I have also tried using Set(CHANNEL(txgain)=x) and Set(CHANNEL(rxgain)=y) and 
those don't seem to have any effect either. I have tried setting x and y to 
-30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no 
change on the phone volume. I can see that the Set(CHANNEL) instruction is 
being executed on the Asterisk CLI.
I am using PJSIP and just upgraded to Asterisk 13.5.0. It wasn't working on 
13.4.0 either, but when I saw the release notes on 13.5 and volume was 
addressed, I was hopeful that it might solve my problem for me.
So I have a couple of questions:
1) Am I using the correct functions in the dial plan to adjust volume? It would 
be something like:same => n,Set(VOLUME(TX)=3)orsame 
=>n,Set(CHANNEL(rxgain)=0)2) If this is correct, what are the min/max values 
that I can use when adjusting volume? I was digging around the source code and 
it looked like maybe a min of -4 and max of +4 was expected - but I am unsure.
If anyone has any ideas on how to do this, I would appreciate any help.
Thanks a lot,
--Matt-- 
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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  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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-- 
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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 
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  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]

that being said: this is one of those cases where the current
behaviour - which is handling an extreme edge case - feels worse than
ignoring that edge case. It's not like we let folks update "core" CDR
values in any case, so you aren't in any danger of changing the billsec on
a forked CDR. The worst that happens is you update the userfield on forked
& closed CDRs when you didn't think it would update, in which case I
suppose you could just use another field. Or read it first and append it
from the dialplan.



> Is there any chance the feature was left out by an accident and if so, is
> there a plan to add it again?
>
>
> My extensions.conf:
> exten => h,1,NoOp(${CDR(userfield)})
> exten => h,n,Set(CDR(userfield)=changed)
> exten => h,n,NoOp(${CDR(userfield)})
> exten => h,n,System(sleep 5)
> exten => h,n,NoOp(${CDR(userfield)})
> exten => 10,1,Set(CDR(userfield)=empty)
> exten => 10,n,Dial(SIP/10)
>
> Detailed log:
> http://pastebin.com/fZ9RAhL4
>
>
>
I'd be fine if you'd like to open an issue for it. If you have 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  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.

-- 
Matthew Jordan
Digium, Inc. | Director of Technology
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 
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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Digium, Inc. | Director of Technology
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
 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

-- 
Matthew Jordan
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  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" 
> To: 
> 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
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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  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.

-- 
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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] Distributed Device States - Best Option

[Matthew Jordan]

On Sat, Jun 27, 2015 at 11:28 AM, Bryant Zimmerman  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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