Re: [asterisk-users] Configuring Opus Forward Error Correction in Asterisk 16 (FreePBX)?

[Kevin H.]

Yes FEC must be enabled through configuration, but be aware that
enabling FEC does not necessarily mean it's being utilized. For
instance, in order for FEC data to be decoded by Sangoma's codec_opus
module for Asterisk several things must occur:

1) it must be enabled through configuration
2) both sides of a call must negotiate for it (via SDP)
3) packet loss must be perceived by the codec_opus module
4) a frame containing FEC data is received

Also, please see the following for more information:

https://wiki.asterisk.org/wiki/display/AST/Codec+Opus

https://wiki.asterisk.org/wiki/display/AST/Asterisk+19+Configuration_codec_opus

https://www.asterisk.org/configuring-opus-encoder-asterisk/

https://www.asterisk.org/asterisk-opus-packet-loss-fec/

- Kevin



On Wed, Jul 20, 2022 at 10:48 AM Brant Merryman 
wrote:

> Hi. I am using Asterisk 16.27.0 in FreePBX 15.0.23.11. I installed via the
> FreePBX ISO (SNG7-PBX-64bit-2104.iso). I used the GUI to enable the Opus
> Codec in Asterisk SIP Settings and I can confirm the calls are using Opus
> 16000. How can I turn on fec. I am wondering if there might be a
> configuration file I need to modify to add “fec=yes” or something like that
> to turn this on.
>
> Thanks in advance for any help
>
> Brant Merryman
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Re: [asterisk-users] Logging different verbosity levels

[Kevin Harwell]

So this turned out more complicated than I originally thought!

My expectation:

Verbosity gets logged using an "at least" check against the current
system's verbose level, which if passed subsequently gets checked against
the logging channel's verbose level. Thus only verbose messages with a
level less than or equal to the system's level AND the channel's level
would be logged to that channel. For example given a system verbose level
of 3, and your setup then the following should occur:

same => n,Verbose(0,Test message verbosity 0) ; gets logged to
logtest.verbose.3, logtest.verbose.2, logtest.verbose.1, and
logtest.verbose.0
same => n,Verbose(1,Test message verbosity 1) ; gets logged to
logtest.verbose.3, logtest.verbose.2, and logtest.verbose.1
same => n,Verbose(2,Test message verbosity 2) ; gets logged to
logtest.verbose.3 and logtest.verbose.2
same => n,Verbose(3,Test message verbosity 3) ; gets logged to
logtest.verbose.3
same => n,Verbose(4,Test message verbosity 4) ; nothing logged
same => n,Verbose(5,Test message verbosity 5) ; nothing logged
same => n,Verbose(6,Test message verbosity 6) ; nothing logged
same => n,Verbose(7,Test message verbosity 7) ; nothing logged
same => n,Verbose(8,Test message verbosity 8) ; nothing logged
same => n,Verbose(9,Test message verbosity 9) ; nothing logged

Reality:

What you saw in your output. As to why? Well it's a bit of a mess and it's
been that way for a while it seems. To start the app_verbose application
gets capped at verbose level 4. Meaning the following:

same => n,Verbose(7,Test message verbosity 7)

Essentially gets changed to:

same => n,Verbose(4,Test message verbosity 7)

Next, Asterisk sets the global verbose logging level according to the
following order with each level potentially overriding if greater than the
preceding:

1) core verbose level (set in asterisk.conf)
2) console level (set via system CLI, or asterisk CLI)
3) channel log level (set in logger.conf, e.g. verbose(9))

This means in asterisk.conf if you set verbose=2, and then on the CLI set
it to 3, and then for any channel in logger.conf specify verbose(9) then
the final level will be 9! The levels too are tracked via different
variables so given the preceding example the following will be output:

*CLI> core show settings
...
Root console verbosity:  2
...

Despite the actual level being 9.

Lastly, when the logger goes to output the actual log message to a channel
it does indeed first check if the channel's verbosity level is greater than
some "level" variable. But guess what? This variable is always equal to 0.
Thus as long as a channel's level is > 0 it always passes and the message
is output.

So you can see in your setup the final system level is indeed 9, and all
app_verbose messages being output are forced to 4 or less. Since the system
level is 9, and the channel's verbosity level is greater than zero then all
messages are output to all files.

Hope that makes sense!

Given all that, in my opinion there seems to be at least one or more bugs
with regards to verbose logging. Please file an issue at
https://issues.asterisk.org/. Feel free to copy/paste what's here as part
of the issue description.


On Mon, May 23, 2022 at 3:29 AM Antony Stone <
antony.st...@asterisk.open.source.it> wrote:

> Hi.
>
> Does no-one else know either?  I thought this was a simple question, and
> it
> was just me being unable to find the appropriate documentation to explain
> how
> these logging levels work.
>
> Please, can anyone help?
>
> On Friday 20 May 2022 at 15:33:45, Antony Stone wrote:
>
> > Hi.
> >
> > I'm trying to use different logging verbosity levels to get dialplan
> output
> > into different log files, and there's clearly something I haven't
> > understood about how Asterisk does this...
> >
> >
> > I have the following in /etc/asterisk/logger.conf:
> >
> > [logfiles]
> > logtest.verbose.0 => verbose(0)
> > logtest.verbose.1 => verbose(1)
> > logtest.verbose.2 => verbose(2)
> > logtest.verbose.3 => verbose(3)
> > logtest.verbose.4 => verbose(4)
> > logtest.verbose.5 => verbose(5)
> > logtest.verbose.6 => verbose(6)
> > logtest.verbose.7 => verbose(7)
> > logtest.verbose.8 => verbose(8)
> > logtest.verbose.9 => verbose(9)
> >
> > I then put the following at a particular point in my dialplan:
> >
> > same => n,Verbose(0,Test message verbosity 0)
> > same => n,Verbose(1,Test message verbosity 1)
> > same => n,Verbose(2,Test message verbosity 2)
> > same => n,Verbose(3,Test message verbosity 3)
> > same => n,Verbose(4,Test message verbosity 4)
> > same => n,Verbose(5,Test message verbosity 5)
> > same => n,Verbose(6,Test message verbosity 6)
> > same => n,Verbose(7,Test message verbosity 7)
> > same => n,Verbose(8,Test message verbosity 8)
> > same => n,Verbose(9,Test message verbosity 9)
> >
> > I was expecting to get each message output into the respective filename,
> > but instead I got 10 files with the expected filenames, and all
> containing
> > every test message, no matter 

Re: [asterisk-users] Setting up sipml5

[Kevin Harwell]

On Fri, Sep 10, 2021 at 12:44 PM Jerry Geis  wrote:

> HI All,
>
> I am trying to get SIPml5 working with 18.6.0.
> My http.conf file:
> enabled=yes
> bindaddr=myip
> bindport=8088
> serverName=MyName
> tlsenabled=true
> tlsbindaddr=myip
> tlscertfile=/etc/letsencrypt/live/mpname/fullchain.pem
>
> The SIPMl log just says:
> WebSocket connection to 'wss://myIP:8088/' failed:
>
> Is there something easy I'm missing to allow websockets on Asterisk ?
> Thanks
>
>
Check out the following wiki pages (if you haven't already), and ensure all
your settings are correct:

https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
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Re: [asterisk-users] Asterisk Getting Crashed

[Kevin Harwell]

On Thu, Jun 25, 2020 at 4:12 PM Ahmed Chohan 
wrote:

> Hi,
>
> Currently I'm experiencing crashes on Asterisk more recently, see messages
> below (crashed reason: segfault signal 6).
>
> abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT
> - dumping core
>
> asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!,
> Failed assertion bad magic number 0x0 for object 0x7fbd2c
>
> 00d170 (0)
>
> After running the backtrace for the coredump, I'm unable to pinpoint the
> root cause of it (see partial messages for the backtrace below).
> Furthermore, I've checked in the forums and advised the "utils.so" module
> issue but I don't think it might be causing this crash.
>
> [root@alpha01 ccpp-2020-06-25-10-46-01-7082]# gdb /usr/sbin/asterisk
> coredump
>
> GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-119.el7
>
> Copyright (C) 2013 Free Software Foundation, Inc.
>
> License GPLv3+: GNU GPL version 3 or later <
> http://gnu.org/licenses/gpl.html>
>
> This is free software: you are free to change and redistribute it.
>
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
>
> and "show warranty" for details.
>
> This GDB was configured as "x86_64-redhat-linux-gnu".
>
> For bug reporting instructions, please see:
>
> <http://www.gnu.org/software/gdb/bugs/>...
>
> Reading symbols from /usr/sbin/asterisk...done.
>
> [New LWP 15373]
>
> [New LWP 15800]
>
> [New LWP 16125]
>
> [New LWP 15829]
>
> [New LWP 16486]
>
> ..
>
> Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
>
> Program terminated with signal 6, Aborted.
>
> #0  0x7fbe7a65f337 in ?? ()
>
> (gdb) bt full
>
> #0  0x7fbe7a65f337 in ?? ()
>
> No symbol table info available.
>
> #1  0x7fbe7a660a28 in ?? ()
>
> No symbol table info available.
>
> #2  0x0020 in ?? ()
>
> No symbol table info available.
>
> #3  0x in ?? ()
>
> No symbol table info available.
>
>
> OS I'm running is CentOs 7.7.1908 and the Asterisk version is 13.21-cert3.
> Please advise.
>
>
Unfortunately debug symbols were not enabled on your system so the
backtrace doesn't have any extractable information. Please see the wiki [3]
on how to get a useful backtrace.

Before that though I recommend upgrading to the latest version of Asterisk
[1]. Or if you're set on using a certified version [3]. The version you are
on is quite old, and there is a decent chance the problem you are
experiencing has been fixed.

[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://downloads.asterisk.org/pub/telephony/asterisk/
[3] https://downloads.asterisk.org/pub/telephony/certified-asterisk/

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Re: [asterisk-users] error compiling current git

[Kevin Harwell]

On Thu, Feb 27, 2020 at 8:51 AM hw  wrote:

> Hi,
>
> compiling the current git version on Centos 7 gives me:
>
>
>[CC] res_statsd.c -> res_statsd.o
> res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified
> in initializer
>   .on_valid_pair = ast_rtp_on_valid_pair,
>   ^
> res_rtp_asterisk.c:2669:2: warning: initialization from incompatible
> pointer type [enabled by default]
> res_rtp_asterisk.c:2669:2: warning: (near initialization for
> ‘ast_rtp_ice_sess_cb.on_ice_complete’) [enabled by default]
>[CC] res_format_attr_g729.c -> res_format_attr_g729.o
>
>
> Is this to be expected or should I make a bug report?
>
>
When you pulled the lasted code this change would have forced a
re-configure. If you haven't already try doing a full clean and rebuild,
and see if you still have the error:

$ make distclean
$ ./configure [your options]
$ make

-- 
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Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

[Kevin Harwell]

On Tue, Feb 25, 2020 at 4:02 PM Patrick Wakano  wrote:

> Hi Kevin!
> Thanks very much for your reply! Much appreciated!
>

You're welcome!


> So I just have a remaining question from this, if the with-ssl is not
> mandatory to have the encryption support, what is it actually used for?
>

In Asterisk is allows you to set a path to the openssl files. Typically
used if your install of openssl is in a non-default alternative location.
For example:

--with-ssl=/path/to/ssl_files

I'm not really sure how Asterisk accepts the parameter without a file path
specified. My guess is it either uses the default path, or potentially sets
some flag meaning ssl is required.


> Maybe it is some old flag which is not needed anymore and so can be
> ignored for now and possibly removed from the configure/makefile stuff for
> future releases?
>
>
See above. The flag is still needed in Asterisk. However, the setting
appears to propagate in some way into the bundled pjproject configuration.
This is in some way affecting the build of pjproject, then subsequently
causing res_pjsip in Asterisk to not load at runtime. Further investigation
is required as to why though.

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Re: [asterisk-users] pjsip startup errors when using "with-ssl" configure option

[Kevin Harwell]

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano  wrote:

> Hello list,
> Hope you are all doing well!
>
> I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
> I wonder if someone can put some light on it.
> Log history short, install_prereq fails to install the packages (not sure
> how important they actually are): speexdsp-devel, gmime-devel,
> uriparser-devel, iksemel-devel, uw-imap-devel, hoard
> Then, I am running the following commands to build Asterisk:
> ./configure --with-crypto --with-srtp --with-ssl
> make menuselect.makeopts
> menuselect/menuselect --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES
> --enable MALLOC_DEBUG --disable BUILD_NATIVE --enable app_macro
> menuselect.makeopt
> make OPT=-fPIC
> make install
> make samples
>
> After this, when I start Asterisk, I get the following error with pjsip
> modules:
> ERROR[6253]: loader.c:2396 load_modules: Error loading module
> 'chan_pjsip.so': /usr/lib/asterisk/modules/chan_pjsip.so: undefined symbol:
> ast_sip_cli_traverse_objects
> ERROR[6253]: loader.c:2396 load_modules: Error loading module
> 'res_pjsip.so': /usr/lib/asterisk/modules/res_pjsip.so: undefined symbol:
> pjsip_tls_transport_start2
> ERROR[6253]: loader.c:2396 load_modules: Error loading module
> 'res_pjsip_config_wizard.so':
> /usr/lib/asterisk/modules/res_pjsip_config_wizard.so: undefined symbol:
> ast_sip_get_sorcery
>
> After a lot of investigation I found this post (
> https://asteriskfaqs.org/2018/09/25/asterisk-users/asterisk-1561-symbol-pjsip_tls_transport_start2-not-found.html)
> which pointed me to the with-ssl parameter.
> So if I run all previous commands, but remove the "--with-ssl" options
> from the configure, then I don't have the pjsip errors when I start
> Asterisk.
>
> My first question is, what will be the impact of removing the --with-ssl
> option? Will TLS and WSS still be possible?
>

If Asterisk is able to autodetect, and find the appropriate encryption
libraries then yes TLS and WSS should be available. I do not use those
options when building and am able to still make secure calls.


> I didn't have time to test these things, however I did compared the
> configure and make outputs and the only difference are below, so looks like
> nothing extra gets compiled when the with-ssl is used...
> With --with-ssl:
> [pjproject]  Configuring with *--enable-ssl *--prefix=/opt/pjproject
> --disable-speex-codec --disable-speex-aec --disable-bcg729
> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec
> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr
> --disable-silk --disable-opus --disable-video --disable-v4l2
> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv
> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc
> --without-external-pa --without-external-srtp --disable-resample
> --disable-g711-codec --enable-epoll
> checking for mandatory modules:  PJPROJECT CRYPTO SRTP *OPENSSL*... ok
> Without --with-ssl:
> [pjproject]  Configuring with --prefix=/opt/pjproject
> --disable-speex-codec --disable-speex-aec --disable-bcg729
> --disable-gsm-codec --disable-ilbc-codec --disable-l16-codec
> --disable-g722-codec --disable-g7221-codec --disable-opencore-amr
> --disable-silk --disable-opus --disable-video --disable-v4l2
> --disable-sound --disable-ext-sound --disable-sdl --disable-libyuv
> --disable-ffmpeg --disable-openh264 --disable-ipp --disable-libwebrtc
> --without-external-pa --without-external-srtp --disable-resample
> --disable-g711-codec --enable-epoll
> checking for mandatory modules:  PJPROJECT CRYPTO SRTP... ok
>
> Also, why am I having the pjsip startup errors when the --with-ssl is
> used? I could not find a clear explanation for this problem and how to fix
> it
>

There appears to be a bug here. I configured, built, and ran with the same
options mentioned (--with-ssl, etc...) and received similar pjsip module
load errors. Please file a bug report on the Asterisk issue tracker [1].

[1] https://issues.asterisk.org/

Thanks!

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Re: [asterisk-users] [asterisk-app-dev] ARI Get Channel Variable

[Kevin Harwell]

On Wed, Jan 22, 2020 at 5:32 PM Phil Mickelson  wrote:

> I'm trying to get the Call-ID from the SIP HEADER using getChannelVar.
> When I pass SIP_HEADER() and anything as the variable I get Unable to read
> provided function.  If use Call-ID I get Provided variable was not found.
>
> This is a connected call.  Is it not possible to get SIP HEADER
> information once it's connected?  Or, am I missing something?
>

Are you using chan_sip or chan_pjsip? If chan_pjsip then you need to use
the PJSIP_HEADER function [1] instead. For instance the following returned
the call-id for me:

PJSIP_HEADER(read,Call-ID)

How are you attempting to use the channel function [2]? For instance the
following returned the call-id for me as well:

CHANNEL(pjsip,call-id)

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_HEADER
[2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CHANNEL

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Re: [asterisk-users] PJSIP Setup Outbound SIP Trunk

[Kevin Harwell]

On Mon, Oct 14, 2019 at 11:56 AM Ahmed Chohan 
wrote:

> Hi,
>
> I've currently migrating from chan_sip to chan_pjsip, for now I'm able to
> setup and configured extensions in PJSIP and incoming trunks but unable to
> configure outbound trunk as getting unauth/unregistered trunk endpoint
> message error message when making outbound calls. However, for inbound
> calls I'm not facing any issues.
>
> I would like to know how can I configured outbound sip trunk bypassing
> registration and auth?
>

Where are the messages coming from? Is Asterisk sending an outbound
registration, but getting rejected? If so make sure your username/password
credentials are correct.


>
> See below current configuration;
>
> [trunk_proxy]
> type=endpoint
> transport=transport-udp
> context=fromsip
> disallow=all
> allow=ulaw
> aors=trunk_proxy
> force_rport=no
> direct_media=yes
> ice_support=no
> trust_id_inbound=yes
> outbound_auth=trunk_proxy
>
> [trunk_proxy]
> type=aor
> contact=sip:10.3.120.208:5060
>
> [trunk_proxy]
> type=identify
> endpoint=trunk_proxy
> match=10.3.120.208
>
> [trunk_proxy]
> type=auth
> auth_type=userpass
> password=
> username=sip_proxy
>
> [trunk_proxy]
> type=registration
> outbound_auth=trunk_proxy
> server_uri=sip:10.3.120.208:5060
> client_uri=sip:10.3.120.208:5060
> auth_rejection_permanent=no
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
> --
> _
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>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Experiencing what I think are issues with the confbridge 'video_mode = follow_talker' and also the talk detection

[Kevin Harwell]

; Value: 4
>
>
>
> Action: SetVar
>
> ActionID: C176
>
> Channel: PJSIP/webrtc_client1-000e
>
> Variable: CONFBRIDGE(bridge,template)
>
> Value: 2
>
>
>
> Action: SetVar
>
> ActionID: C177
>
> Channel: PJSIP/webrtc_client1-000e
>
> Variable: CONFBRIDGE(user,template)
>
> Value: 4
>
>
>
> [2]
>
> type = bridge
>
> language = en
>
> internal_sample_rate = 0
>
> mixing_interval = 20
>
> record_file_append = no
>
> max_members = 10
>
> video_mode = follow_talker
>
>
>
> [4]
>
> type = user
>
> admin = no
>
> marked = no
>
> startmuted = no
>
> music_on_hold_when_empty = no
>
> quiet = yes
>
> wait_marked = no
>
> end_marked = no
>
> dsp_drop_silence = yes
>
> dsp_silence_threshold = 2500
>
> dsp_talking_threshold = 160
>
> denoise = no
>
> jitterbuffer = yes
>
> talk_detection_events = yes
>
> dtmf_passthrough = no
>
> announce_user_count = no
>
> announce_join_leave = no
>
> announce_user_count_all = no
>
> announce_only_user = no
>
> send_events = no
>
> echo_events = no
>
> announce_join_leave_review = no
> --
>
>

[1] https://issues.asterisk.org/
[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Thanks!

-- 
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Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video

[Kevin Harwell]

On Wed, Mar 13, 2019 at 10:15 AM Dan Cropp  wrote:

> Using asterisk 16.1.1.
>
>
>
> I’m setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate
> Dynamic Edition).
>
>
>
> I have noticed Chrome 72 had some issues with video streams.  I just
> upgraded to Chrome 73 and see they still have some issues.  If I have 2
> calls in a confbridge with video set to none.  I then set the video source
> to a Chrome browser and the Remote Video shown to both calls from Firefox
> and Chrome do not update.  However, if I set the video source to the
> Firefox browser, my Remote Video is accurate in both Firefox and Chrome.
>
> I confirmed that asterisk is indicating the video source changed by
> looking at the AMI BridgeVideoSourceUpdate event.
>
>
>
> When I use the Firefox (65.0.2) browser I can set either call to be the
> video source and the Remote Video updates accordingly.
>
>
>
> Is this caused by Chrome’s video sent to asterisk being some format which
> asterisk can’t use in the confbridge?
>

The way I understand it is that video stream has changed, so the browser
needs some way to know that. Otherwise the decoder thinks it's invalid data
and drops it. In these cases either Asterisk needs to issue a renegotiation
(currently not supported), or the codec needs to contain video stream
information in their payload. I spoke with Joshua Colp about this some as
he's had some dealings with this and he had the following to say:

"Some codecs (such as VP8/V9) embed information about the video stream
within their payload. Asterisk does not currently rewrite this information,
so when a stream change occurs in Asterisk using the selective source
functionality this can cause the receiving side (the browser) to drop the
payload as it sees it as not being part of the existing stream. Different
browsers can behave differently, such as resetting the video decoder to handle
the new stream. Rewriting this information in the video payload is not
currently supported."

So you are probably seeing it work or not in Chrome vs Firefox due to
browser, and codec support of such occurrences.

     


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Re: [asterisk-users] Question on WebRTC configuration

[Kevin Harwell]

On Fri, Dec 7, 2018 at 9:11 AM Dan Cropp  wrote:

> In the asterisk wiki instructions for Configuring Asterisk for WebRTC
> clients…
>
>
>
>
> https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
>
>
>
> “To communicate with websocket clients, Asterisk uses its built-in HTTP
> daemon.  Configure */etc/asterisk/http.conf* as follows:
>
>
>
> [general]
>
> enabled=yes
>
> bindaddr=0.0.0.0
>
> bindport=8088
>
> tlsenable=yes
>
> tlsbindaddr=0.0.0.0:8089
>
> tlscertfile=
>
> tlsprivatekey=
>
> tlscafile=”
>
>
>
> What is the tlscafile setting?
>
>
>
> When I look at the http.conf samples it doesn’t mention the tlscafile
> setting.
>
> I see there is a tlscafile setting in sip.conf, but I don’t find this
> anywhere else.
>
>
>
> Is the wiki web page mistaken or is this an actual http.conf setting that
> is undocumented?
>

The page is mistaken. It should not be there. the 'tlscafile' option is not
supported by the Asterisk http server. I've removed it from the wiki.
Thanks for catching that!


>
>
> Have a great day!
>

You too!


> Dan
> --
>

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Re: [asterisk-users] SIPp scenario file for testing UAC Authentication with Asterisk ?

[Kevin Harwell]

On Thu, Oct 25, 2018 at 4:32 AM Olivier  wrote:

> Hello,
>
> I'm testing an Asterisk instance.
> At the moment, I'm focusing on its capability to receive and challenge
> incoming SIP Registrations.
>
>
If all you want to do is test inbound registrations you can find an example
SIPp scenario in the Asterisk testsuite[1]. You'll want to remove the
 section from the 200 response and the variable reference. Then
you'll want to execute the scenario with something like the following
(replacing with your values of course):

sipp  -m 1 -sf register-auth.xml -s  -ap 

Another example can be found here[2]. Both the README and register.xml file
have instructions on how to execute the test. Currently the test is setup
to test a few thousand endpoints though. However you can adjust that number
by modifying the register.csv (injection file) or by not using the
injection file and modifying the register.xml scenario itself.

[1]
https://github.com/asterisk/testsuite/blob/master/tests/channels/pjsip/registration/inbound/nominal/single_contact/authed/sipp/register-auth.xml
[2]
http://blogs.asterisk.org/wp-content/uploads/2018/09/performance_inbound_registration.tar.gz



> For various reasons, I would prefer to use SIPp instead of Asterisk to act
> as SIP Client.
>
> Has someone successfully done this ?
> If negative, what explains this ?
> If positive, can you give an example of a successful SIPp scenario file ?
> I've played with both embeded branchc and [1] but met no success yet
>
> Best regards
>
> [1] https://github.com/rkday/sipp-samples/blob/master/uac-auth.xml
>
>

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[asterisk-users] pjsip trunk config question + DNS related error messages

[Kevin Long]

Greetings,


I am getting the following error (below) continually in my asterisk log, 
related to qualify_frequency I believe. I am trying to use sip trunking with 
the company flowroute.

3 questions if I may:

1) Is using qualify_frequency with a sip trunk a common or recommended 
practice? I figured it would function as a keep-alive and keep the ‘pjsip show 
endpoints’ availability data up-to-date if I wanted to check on the health of 
the trunk. Sound right?

2) Any idea what this error means? Googling showed almost nothing except one 
other post to this list for a bug that was fixed in 14 , I’m on 15.x

3) Any other recommendations for this trunking config?


Thanks very much ! Especially jcolp and gtjoseph for answering my queries in 
the past, sorry if I don’t always respond again as I haven’t actually figured 
out a good way to do that unless I am subscribed to receiving all mails from 
the list.




[Mar 28 23:17:43] ERROR[4812]: res_pjsip.c:3770 endpt_send_request: Error 
320047 'No answer record in the DNS resp
onse (PJLIB_UTIL_EDNSNOANSWERREC)' sending OPTIONS request to endpoint flowroute


[flowroute]
type=auth
auth_type=userpass
password=**
username=**
[flowroute]
type=aor
contact=sip:sip.flowroute.com:5060
qualify_frequency = 15

[flowroute]
type=endpoint
transport=transport-udp
context=from-flowroute
disallow=all
allow=ulaw
outbound_auth=flowroute
aors=flowroute

[flowroute]
type=identify
endpoint=flowroute
match=216.115.69.144
match=70.167.153.130

  1.
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[asterisk-users] how to get "SMS" messages (http) into Asterisk "sip messages"

[Kevin Long]

Hello,

We are building a shim to get SMS messages (which come in from twilio via an 
http post to our python web app), forwarded on to the appropriate SIP client 
registered to asterisk.

The application receiving the “SMS” via HTTPS from twilio does not have a SIP 
component.

I am hoping there are different ways to get the message details into Asterisk 
so that it can create a MESSAGE and send it to the local endpoint.

Does anyone know the best way to get this information into Asterisk? Can I do 
it with AMI, AGI, a file queue ?

Would love to hear from anyone who has implemented something like this. 
Outbound is the easy part. How are you handling inbound SMS->SIP ?


Regards,

Kevin Long


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[asterisk-users] pjsip trunking configuration issue

[Kevin Long]

Greetings ! 


My goal is to get Twilio trunking working, and with TLS/SRTP. 

I see this concerning message in my log:

[Feb  7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object 
of type 'endpoint' with id ’twilio' from configuration file ‘pjsip.conf’



Thus, ‘pjsip show endpoints’  does not show the endpoint for the Twilio trunk. 


Hoping for a sanity check of my pjsip.conf file, and what could be causing 
this.  

A test call form Twilio’s system hits the PBX (over TLS), but always says “No 
matching endpoint found” in the asterisk log.



pjsip.conf

[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
cert_file=cert_file
priv_key_file=key_file
method=tlsv1
external_media_address=X.Y.Z.D
external_signaling_address=X.Y.Z.D
verify_client=no
verify_server=no
allow_reload=yes

[twilio](!)
type=endpoint
transport=transport-tls
context=from-twilio
disallow=all
allow=ulaw
dtmf_mode=inband
media_encryption=sdes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
canreinvite=no
tlsdontverifyserver=yes


[auth-out](!)
type=auth
auth_type=userpass

[twilio]
aors=twilio-aors

[twilio-aors]
type=aor
contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also

[twilio]
type=identify
endpoint=twilio
match=54.172.60.0
match=54.172.60.1
match=54.172.60.2
match=54.172.60.3

[endpoint-basic](!)
type=endpoint
transport=transport-tls
context=from-phones
disallow=all
allow=ulaw

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

[aor-single-reg](!)
type=aor
max_contacts=20

[1001](endpoint-basic)
auth=auth1001
aors=1001

[auth1001](auth-userpass)
password=password123
username=1001

[1001](aor-single-reg)


Extensions.conf

[from-twilio]
exten => _+1NX,1,Dial(PJSIP/1001)

[from-phones]
exten => _NXXNXX,1,Set(CALLERID(all)="David" <78451234>)
same => n,Dial(PJSIP/+1${EXTEN}@twilio)
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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning Mon 01/08/2018)

[kevin . larsen]

I am out of the office from Thu 01/04/2018 until Mon 01/08/2018.

I am out of the office and will have limited contact. For all
emergencies/issues, please contact the helpdesk at
helpd...@pioneerballoon.com or 316-688-8777.


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Duplicate CDR's in mysql" sent on 1/4/2018 12:44:33 PM.

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Re: [asterisk-users] Rewrite Outgoing Number

[kevin . larsen]

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:52:32 AM:

> From: "basti" 
> To: asterisk-users@lists.digium.com
> Date: 12/14/2017 09:52 AM
> Subject: Re: [asterisk-users] Rewrite Outgoing Number
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> ok thanks for the answer, i will try it.
> sorry for the question: in which file should it be configured?
> 

In FreePBX, you will set up provider 1 as an outbound route 
(Connectivity/Outbound Routes). You will tell it what dial patterns to use 
that will take that route. You will also specify which trunks to use and 
in what order they should go. One would assume in your situation that you 
want to have provider 1 as your primary trunk and provider 2 as your 
backup trunk should trunk 1 be down.

So, basically, you first need to set up provider 1 and provider 2 under 
Connectivity/Trunks. Make sure that under CID Options you have Allow Any 
CID, otherwise your test won't work. Then you need to set up outbound 
routes under Connectivity/Outbound Routes. Make sure there that Your trunk 
sequence has the Provider 1 trunk as primary. If you want Provider 2 as a 
backup, put that as secondary.

Finally, make sure that under Applications/Extensions, on the General tab 
that you have the Outbound CID set to the number you want to use. That 
will get used whenever you dial out a trunk.

Hope that sets you down the correct path.

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Re: [asterisk-users] Rewrite Outgoing Number

[kevin . larsen]

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208

asterisk-users-boun...@lists.digium.com wrote on 12/14/2017 09:36:06 AM:

> From: "basti" <mailingl...@unix-solution.de>
> To: asterisk-users@lists.digium.com
> Date: 12/14/2017 09:36 AM
> Subject: Re: [asterisk-users] Rewrite Outgoing Number
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> On 14.12.2017 16:30, basti wrote:
> Hello,
> I am new on asterisk and do some tests on freepbx.
> 
> I have 2 SIP provider:
> 
> Provider1: In-/Out- Flatrate, only 1 Number
> Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 
numbers
> 
> On Asterisk site i have 3 phones
> (branch ??, don't know how its called in asterisk)
> 
> Is it possible to do something like:
> 
> Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: 
Number1/Provider1
> Phone 2: Incoming Call: Number1/Provider2 Outgoing Call: 
Number1/Provider1
> Phone 3: Incoming Call: Number2/Provider2 Outgoing Call: 
Number1/Provider1
> 
> I have forgotten an essential thing:
> 
> Phone2 und Phone 3 should use  Line Number1/Provider1 for Outgoing Call
> but show Number1/Provider2 or Number2/Provider2 on caller side.

If, and this is a big if, your provider 1 allows you to use a caller ID 
number that they do not control, then yes, you can do what you want. 

Some providers allow this and some do not. It may be that provider one 
will overwrite whatever you set as caller ID with the number you have 
purchased from them. It may also be that they will allow you to set a 
different outbound caller id. Also, the person receiving the call will not 
know if you have provider 1 or provider 2. It is purely the number and 
possibly a name that they will see.

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 07/31/2017)

[kevin . larsen]

I am out of the office until 07/31/2017.

I am out of the office and will have limited contact. For all
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[asterisk-users] RTP / NAT question with IPv6/IPv4 problem

[Kevin Long]

Hello,

All my asterisk systems use only IPv4 currently.  I have one phone which is on 
T-Mobile network,  and this network is only IPv6 now.  

The phone can register fine, because T-Mobile does NAT64 and it connects fine 
to my IPv4 asterisk server. 

But in the SDP for a call setup,  this phone sends only an IPv6 address as a 
contact, so RTP fails.


I have nat=yes already set on this chan_sip extension, I thought this would 
ignore the IPv6 in the SDP and use the *apparent* IPv4 instead, but apparently 
not?

Any help appreciated, thanks all.



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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> I've already proposed your solution (is the most reasonable) but they
> have more than 60 analogs lines (no faxes) and some of them terminate in
> appliances like alarms, etc, so the solution must not touch in any way
> the connection between the line and his termination: doing a analog to
> digital conversion, passing it to asterisk and the convert it back to
> analog is prone to problems (what if asterisk crashes? or if a gateway
> fail?).
> I can split the existing lines (there are no complex things like adsl or
> digital signaling), convert the branches to digital and terminate then
> into an asterisk machine, so any failure will not affect the old
> circuit, but of course I've to configure asterisk to ONLY LOG calls and
> nothing more.
> 
> This is what they want:
> - line 1 ring
> - line 1 is splitted in two, the first branch (let's say the "analog"
> branch) go to an analog phone, that rings
> - the second branch go through a gateway and then to asterisk
> - asterisk log (with an AGI for example) "line 1 rings at  from 
"
> no more is required from asterisk, if someone answer the analog phone or
> not is not my business.
> 
Ok, so I would agree with them that a conversion to digital and back again 
would tend to break things like fax lines and alarm lines. My analog lines 
in my facilities are there because a lot of alarm systems just don't work 
with SIP at all. It's something the alarm companies are going to have to 
figure out in the next decade or so as the Telcos are moving away from 
copper and switched networks and towards fiber and packet based networks.

I honestly don't know if you can do what you want without some piece of 
equipment picking up the line. What I would do is get an analog line, an 
analog phone, an analog to sip device (there are many to choose from) and 
a basic asterisk instance. I would then make a small test setup where the 
analog line goes to a splitter. One side of the splitter goes to your 
analog phone. One side goes to your analog to SIP converter and then into 
your asterisk instance via your ethernet network. Use your cell phone to 
call the number of your analog line and see if it works. You would have to 
code a basic dialplan on the asterisk side and set up the trunk to your 
converter, which I am assuming you know how to do.

This would at least give you a fairly low cost way to test to see if you 
can trigger what you want on the Asterisk side without also triggering the 
line itself to be answered. I would also note that you would only be able 
to log incoming calls this way. I can't see a way you would be able to 
detect an outgoing call from the analog extension.

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Re: [asterisk-users] log incoming calls without answering

[kevin . larsen]

> From: Fabio Moretti 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Date: 04/20/2017 03:26 PM
> Subject: [asterisk-users] log incoming calls without answering
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hi,
> 
> I've some analogic lines and I'm asked if it's possible to program 
> an asterisk for "checking" the inbound calls without answering them,
> doing something like this:
> 
> analog line 1 -+-- asterisk
>|
>\__ analog phone
> 
> when a call enter, asterisk sense it and store its values (callerid,
> date and time, etc) somewhere, but nothing more, people will answer 
> using the old analog phone.
> The goal is to have a log of the inbound calls without touching the 
> old analog system (it's shared between different subjects).
> 
> I'm pretty sure it's something possible, but how to tell asterisk: 
> "ok, call this AGI, and then don't answer and do nothing more".
> 
> Any idea?
> 
> Thanks

This gets kinda Rube Golberg-ish, but convert the incoming analog line to 
sip, route it through asterisk and have asterisk do its thing before 
converting it back to analog to send to the phone. Only problem is you get 
a lot of extra hardware involved in the mix to make it work. It will be a 
lot of expense and trouble, so you need to make sure that whatever part 
you want asterisk to play is worth that effort. Also, I wouldn't touch a 
fax line in this manner.

If you could give a bit more info on what you want asterisk to do, we 
could maybe give better advice on how to solve your problem.

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Re: [asterisk-users] Crashes in jitterbuffer with framedata->timer_interval > 1000

[Kevin Harwell]

On Tue, Apr 18, 2017 at 1:59 PM, Richard Kenner <ken...@gnat.com> wrote:

> I had three crashes this morning on a divide-by-zero, for example at
> abstract_jb.c:1008 in 14.3.0.
>
>
This is quite odd. I took a quick look at the code at that line number and
it appears the divider should never be zero, but somehow it got set to or
initialized to that maybe.


> Does this ring any bell to anybody?
>
>
This does not sound familiar. Please create an issue on the Asterisk issue
tracker [1] and attach a backtrace [2]. Debug logs around the time of the
crash may be helpful too if you have those. Which channel type (chan_sip,
local channel, chan_pjsip) is involved, and how you are enabling the jitter
buffer (dialplan function vs configuration) would be good to know as well.

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

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

[Kevin Long]

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?

Thanks!

Kevin Long
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[asterisk-users] 256 bit SRTP ciphers in Asterisk 14.x , only works for outbound call ?

[Kevin Long]

Greetings,


If I understand correctly,  Asterisk 14 introduced support for some new SRTP 
ciphers (including some 256 bit ones), previously only two 128 bit ciphers were 
supported.

Using Asterisk 14, I was able to make a call from a softphone (Groundwire) with 
a 256 bit cipher suite on SRTP, which is great. 


However,  I don’t see any way to specify with PJSIP (or chan sip) ,  the cipher 
suite which should be used when Asterisk calls the endpoint/phone.


So I believe this means Asterisk would always use 128 bit SRTP to call the 
phone.  Then, if you have 256 bit only ciphers allowed on the phone, the call 
fails.


Perhaps this is just not documented, or may not be implemented yet.  Anyone 
have a thought? 

Thank you,.

Kevin Long
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[asterisk-users] TLS certificate warnings in softphone, but not until after successful registration and call placed ?

[Kevin Long]

Hello,

I am using asterisk 14.2 and PJSIP,  with TLS transport.

I’m sure I’m doing something wrong here .. 


In 2 distinct softphone clients (Bria and Groundwire),  I am able to register 
successfully,  and place a SIP call, with no certificate warnings. But shortly 
after I place that first call and hang up,  I receive a certificate name 
mismatch error in the softphone,  the error presenting me with the *IP 
adddress* of my Asterisk server,  not the hostname, and of course the TLS 
certificates only have the hostname, not the IP, and I have configured the soft 
phone to use the hostname, not the IP, to connect.


I’m guessing there is some currently unset hostname setting within 
asterisk/pjsip that is defaulting to sending the IP in the sip messages,  and 
then when the soft phone tries to make a new tls sip connection to asterisk,  
perhaps to signal to asterisk that the call is complete,  it then connects to 
the IP instead of the hostname, and the mismatch occurs ?


Any help appreciated,

Thanks,

-Kevin


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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[kevin . larsen]

> True agree, problem is somehow the people purchased am 
> supporting to overcome that. Trying level best... around 20 
> phones has been purchased

Ah, yes, the "we purchased these without consulting you, but it is up to 
you to make them work" school of thought. It often goes with, "Well, what 
are we paying you for?" and "It's a phone, it shouldn't take you long to 
make it work."

I have to say, unless I am working with a Cisco phone system, Cisco phones 
are not my favorite beasts to work with.

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Re: [asterisk-users] PJSIP missing objects

[Kevin Harwell]

On Fri, Dec 2, 2016 at 1:50 PM, Saint Michael <vene...@gmail.com> wrote:

> In version 13.13.0
>  there is no
> res_pjsip_keepalive.so
> res_pjsip_multihomed.so
>
> Is this normal?
>

In this case yes. res_pjsip_keepalive was renamed
to res_pjsip_transport_management and res_pjsip_multihomed was removed as
the bulk of its code was moved into the res_pjsip core.

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[asterisk-users] Specify "name" for Resource in RLS

[Kevin Miller]

Is there are way to specify the display name of a resource in a resource
list?  I have setup a resource list in Asterisk 13 for 1234.  All is working
on the device, but I want to show "Joe User" instead of "1234".  Any
thoughts?


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Re: [asterisk-users] FAX CNG detected but no fax extension

[kevin . larsen]

> Hello,
> I have a question regarding incoming fax to local file (on the 
> Asterisk server).
> While the fax is received properly (I have the tiff file generated 
> as expected) I get the warning 'FAX CNG detected but no fax 
> extension' on the consol.
> 
> If the fax is received ok then what 'fax extension' does it expect 
> and what should I do there? 
> 
> My Setup:
> Sender -> Public PSTN -> provider -> SIP trunk (configured with 
> G711a) -> Asterisk (13.6.0)
> 
> My extension.conf on relevant section is this (obviously this is not
> production code):
> exten => s,1,Answer()
> same =>  n,Verbose(0, Attempt to Receive FAX)
> same =>  n,Set(FAXOPT(gateway)=no)
> same =>  n,ReceiveFax(/var/workspace/testfax.tiff,d)
> same =>  n,Hangup()
> 
> and 
> Server*CLI> module show like fax
> Module Description 
> Use Count  Status  Support Level
> res_fax.so Generic FAX Applications 
> 1  Running  core
> res_fax_spandsp.so Spandsp G.711 and T.38 FAX 
> Technologies  0  Running  extended
> 2 modules loaded

The good news is you don't really have anything wrong and as things are 
working as expected, you can ignore the warning if you so choose.

What generates that error is that on your trunk, you have faxdetect=yes. 
This will cause Asterisk to listen in to all your calls on that trunk and 
try to detect a fax and if it finds it will redirect it to a fax extension 
to be handled as a fax.

You have written a fax handler for your fax lines, but that doesn't stop 
the fax detection from trying to route it to an extension called fax. 
Since this doesn't exist in your case, you get the warning, but the fax is 
received because you are handling in the current path.

Where things would actually break is if someone sent a fax to one of your 
voice lines. If you don't have a fax extension to send it to, the person 
being called would pick up to fax tones. If you do have a fax extension, 
they would get the call yanked from them and it would be sent over to the 
fax extension. In my particular case, testing shows I get about half a 
ring to my desk phone before the system determines fax call and sends it 
to the fax system.

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Re: [asterisk-users] Problem "re-parking" calls

[kevin . larsen]

> All;
> I have a problem with regards to “re-parking” calls and I was 
> hoping someone could shed some light on the topic. Consider this 
scenario:
> 
> (1) An inbound call comes in and the attendant answers it
> (2) The attendant places the call on hold and the caller is sent to 
> extension 701
> (3) Blah, blah, blah. The attendant does something and tells John 
> Doe to pick up the call on extension 701
> (4) The attendant then picks up the call on 701 and tells the person
> that John Doe will be right there to help them
> (5)  The attendant then re-parks the call but now the caller is sent to 
702
> (6) John Doe can't find the call anymore
> 
> 
> Is there something obvious that I am missing? Has anyone else found 
> this to be a problem? Any insight at all would be greatly appreciated.
> Regards;
> John V.

Your problem occurs in step 4 & 5. I don't believe that you can pick up 
the call and then ever be guaranteed to get the same parking position when 
you put it back in park. What would happen if someone else parked a call 
in between steps 4 and 5 and they got 701 because it was free. Once 
parked, the call should remain so until it is picked up or times out back 
to the attendant.

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Re: [asterisk-users] Asterisk inside network. What phone works well?

[kevin . larsen]

> I have Asterisk running well inside our network. I did some 
> experiments exposing it to internet but had some issues:
> 1. NAT issues (voice one way, etc). From what I understand double-
> NAT users will always have something like this
> 2. Immediately I see people trying to hack into. I did configure 
> Fail2Ban and it works somewhat, but not 100%. Erroneous logs, etc
> 
> So.. I ended up closing network. Currently most users inside 
> network. My home router have GRE tunnel to office so phone works just 
fine.
> Another user uses VPN and soft phone and it works good too.
> 
> Now I need to setup some users with actual phone devices and none of
> those solutions will work. So, I did some research and found 
> that some phones have VPN capability built in. 
> 
> Right now I use Cisco SPA504G phones. We have auto-provisioning for 
> them, works well. But I don’t think they have VPN capability.
> 
> 
> What I found it that Cisco 525g2 has AnyConnect functionality (SSL 
> VPN) but not sure if this is what I need.
> 
> We have Mikrotik router. Can I setup VPN on router and have this 
> Cisco phone auto-dial VPN and then connect to Asterisk? I’m asking 
> to see if this will work before I go in and buy that phone.
> Or maybe there is other devices/solutions you suggest? I’d like to 
> stay with Cisco because I’m somewhat familiar with provisioning those..

I haven't done this myself, but I think what you need to look at is phones 
that can do IPSEC vpn setups.

For the Mikrotik router, this may be helpful to start investigating:
http://wiki.mikrotik.com/wiki/L2TP_%2B_IPSEC_between_Mikrotik_router_and_a_PC

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 09/06/2016)

[Kevin Larsen]

I am out of the office until 09/06/2016.

I am out of the office and will have limited contact. For all
emergencies/issues, please contact the helpdesk at
helpd...@pioneerballoon.com or 316-688-8777.


Note: This is an automated response to your message  "Re: [asterisk-users]
Need ISDN call generator" sent on 8/29/2016 2:58:18 AM.

This is the only notification you will receive while this person is away.


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Re: [asterisk-users] Getting better Caller ID

[Kevin Larsen]

> Hello,
> 
> We use Asterisk and as per book we use MAC addresses as user names.
> So, when call coming in from outside (SIP trunk) - caller id is good.
> 
> But when users calling each other on extensions - they see MAC 
> addresses. How would I make it so we see actual names instead of MAC
> addresses? Without changing users..
> 
Do you have a line like the following in your sip.conf for each user?

callerid="Name Here" 

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[asterisk-users] PJSIP/Realtime RLS

[Kevin Miller]

I see that you can configure RLS in pjsip.conf, but does this work with
realtime?  The wiki refers to pjsip.conf for configuration, but since many
of the other items can be in the the DB, I was wondering if RLS can as
well.

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[asterisk-users] Asterisk 13 with LDAP ? (single sign on )

[Kevin Long]

Is it possible to configure Asterisk such that numerical extensions and/or 
usernames,   would be populated from LDAP,  as well as authenticate the 
endpoints where the “SIP secret” is equal to the user’s hashed password in LDAP?


I’d like to use LDAP for single-signon as I do with a number of other 
applications,  and am curious if anyone has a working example or if this is 
even possible?


Thank you,

Kevin Long




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Re: [asterisk-users] Need stronger SRTP ciphers (256 bit)

[Kevin Long]

Some more information (would love some thoughts on this, I have never submitted 
a patch yet).

Groundwire (Popular SIP app) supports the following cipher suites for SRTP:


AES_CM_128_HMAC_SHA1_32
AES_CM_128_HMAC_SHA1_80
AES_CM_192_HMAC_SHA1_32
AES_CM_192_HMAC_SHA1_80
AES_CM_256_HMAC_SHA1_32
AES_CM_256_HMAC_SHA1_80
AEAD_AES_128_GCM
AEAD_AES_256_GCM



I see in the asterisk 13.9.1 source tarsal,  in res/res_srtp.c :


Could adding support for the above cipher suites be as simple as adding more 
options to this switch/case statement with the appropriate parameters or is 
there more to it? 

Thank you!



static int policy_set_suite(crypto_policy_t *p, enum ast_srtp_suite suite)
{
switch (suite) {
case AST_AES_CM_128_HMAC_SHA1_80:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 10;
p->sec_serv = sec_serv_conf_and_auth;
return 0;

case AST_AES_CM_128_HMAC_SHA1_32:
p->cipher_type = AES_128_ICM;
p->cipher_key_len = 30;
p->auth_type = HMAC_SHA1;
p->auth_key_len = 20;
p->auth_tag_len = 4;
p->sec_serv = sec_serv_conf_and_auth;
return 0;

default:
ast_log(LOG_ERROR, "Invalid crypto suite: %u\n", suite);








> On May 30, 2016, at 11:49 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote:
> 
> 
> 
> Hi folks,
> 
> 
> At least several endpoints (soft phone and desk phones) are supporting 
> various 256 bit ciphers for SRTP these days.   I *believe* libsrtp has been 
> updated to allow this,   and that only the code in Asterisk has not been been 
> updated to allow these stronger ciphers.
> 
> Would anyone with the know-how be willing/able to submit a patch ?
> 
> 
> Thank you, 
> 
> Kevin Long
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[asterisk-users] Need stronger SRTP ciphers (256 bit)

[Kevin Long]

Hi folks,


At least several endpoints (soft phone and desk phones) are supporting various 
256 bit ciphers for SRTP these days.   I *believe* libsrtp has been updated to 
allow this,   and that only the code in Asterisk has not been been updated to 
allow these stronger ciphers.

Would anyone with the know-how be willing/able to submit a patch ?


Thank you, 

Kevin Long
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Re: [asterisk-users] open source pbx free

[Kevin Larsen]

> Anyone have any experience running an open source pbx and call 
> center solution?Need to start a call center of 10 users  and i need help 
 
> 
> I have already  installer a server with Ubuntu Server 14.04  , E1 
installed 
> 
> Please advice me how to process  from here 
> 
> Regards 
> 
> Yves 

Many of us on this list have experience running call centers off of 
Asterisk, myself included. If you haven't done Asterisk before, you might 
want to bring in some outside help in order to smooth over the process. It 
isn't that you can't do it on your own, but expect there to be something 
of a steep learning curve. If you haven't had experience with VOIP before, 
you will run into issues that you didn't even know were possible, and in a 
call center scenario, you will have people breathing down your neck 
wanting things fixed/changed.

The great thing about Asterisk is that if you know what you are doing, you 
can pretty much bend it to your will. It isn't perfect (no software is), 
but there have been very few requests from end users that I haven't been 
able to fulfill once I understood what they really wanted. Phone systems 
are big and scary and hard for technical people. Most non-techies don't 
know enough about them to even know the right questions to ask. That's why 
your very first job is to find out what does the client really want/need 
their phone system to do. Call center of 10 users gives you a direction to 
go in, but it isn't enough to design the phone system. You need to find 
out what exactly do they want to happen when a call comes in. How should 
it be routed. Are they going to use call queues? By indicating a call 
center it is likely they will, but I have seen it where they don't.

Once you have your requirements mostly decided, then you can go ahead and 
decide on what to do next. If it will fit the bill, especially for a new 
asterisk user, there are many prebuilt distributions that will make 
setting up and maintaining your Asterisk solution easier. They have nice 
web interfaces to handle all the heavy lifting. As you sound pretty new to 
VOIP, this may be the way you want to go. If they don't meet your needs, 
then you may be into custom programming the dialplan and gets a lot more 
involved. 

Good luck and enjoy the journey. Also, the more specific you can make your 
questions, the better and more likely the fine folks on this board will be 
to respond with helpful information.

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Re: [asterisk-users] Recommendations for free virtual server tech and Asterisk?

[Kevin Long]

Personally I am about to try asterisk on proxmox using containers since they 
run code "native". I've had timing issues on conference calls (stutter) with 
VMware esxi . Not sure about KVM I hope it's also better than esxi too.

Sent from my iPhone

> On Apr 6, 2016, at 9:13 AM, Markos Vakondios  wrote:
> 
> Proxmox and KVM on Ubuntu 
> 
>> On Wednesday, 6 April 2016, Ryan, Travis  wrote:
>> What is the best virtual server tech (and most stable, etc) to use for a 
>> asterisk virtual hosting environment?
>> 
>>  
>> 
>> I have a client that wants to do virtual hosting of Asterisk (only SIP or 
>> IAX, no PRI, etc) and I’m wondering if Xen or something else would be best? 
>> We’d like to stay away from the costs of VMWare if possible.
>> 
>>  
>> 
>> Thanks!
>> 
>>  
>> 
>> Travis
>> 
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Re: [asterisk-users] Is possible to use FXO Digium card like a Fax modem?

[Kevin Larsen]

> There are also cheap USB fax modems that you can attach to an FXO 
> port and that works fine. All you have to do then is configure 
> asterisk to detect incoming faxes and route them to that port 
> (faxdetect=yes?).
> 
> This worked great for me when I had all my incoming calls coming 
> over a Century Link POTS line. As I approach retirement and want to 
> save money, I switched from the $44/month POTS line to a pennies-
> per-month VOIP service via IAX registration. So now I'm wondering 
> whether this setup would still work. The question undoubtedly shows 
> my ignorance of telephony stuff. I'm willing to do my homework, I 
> just want to know if it's even possible to do this, or if there are 
> better ways to handle fax over VOIP.

I am going to say this with tongue only partially in cheek. The better way 
to do fax over VOIP is not to do it. It is finicky and unless you have a 
real need for it, it isn't worth the time it takes to make it all work. 
Even working, you still have complaints every time a fax fails to send or 
receive as people somehow have this expectation that faxes should never 
fail. To quote the movie War Games, "The only winning move is not to 
play."

It would be preferable to use a scanner and email to send documents if at 
all possible. If you still need the occasional fax, I would recommend 
using a fax service and letting that be someone else's headache.

That said, my company still has plenty of people who insist that faxes are 
the greatest thing since sliced bread, so I get the fun of supporting 
them. Your options, depending on scale are to use one the solutions you 
can integrate right into the Asterisk server or to use an external package 
and then you just forward the calls from your asterisk box over to your 
fax software (this is the one I use).

Make sure that your SIP/IAX provider supports T.38 faxing (specifically 
transcoding) as this will make your life much easier. You have to be 
careful here as many providers will happily pass T.38 along if it comes in 
that way, but if someone with an analog line/fax setup sends you a fax, it 
will hit their system as audio and pass on to you as audio, which with SIP 
can be fraught with danger unless you have a really excellent connection 
to your sip provider. With transcoding, they can convert it as it enters 
their system to T.38 and then just pass the T.38 to you, which results in 
greater successes. T.38 passthrough is common, transcoding less so, but it 
is getting more common as time goes on.

Also, if your provider does not support T.38 transcoding, plan on sticking 
with ulaw or alaw for faxing. The compressed codecs do not allow the audio 
signal to pass properly and faxes will not work. 

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[asterisk-users] Client TLS certificates for auth ?

[Kevin Long]

I use TLS and SRTP on my Asterisk servers. The server certificates are signed 
by my internal CA, and the Root CA cert is distributed to the phones and soft 
phones so they will trust the server without warning. 

It is not clear to me if Asterisk can be configured to actually reject client 
connections/registrations from peers which do not possess a client certificate 
which has been signed by a particular CA ?

If so, could it be such that the common name in the client certificate would 
need to match the username or Asterisk “extension” ?


I’m wondering if this can be done ,  to have a second factor of authentication 
besides the SIP secret , since in my current setup, despite using a TLS/SSL 
cert for the server, the server only verifies the client by the SIP secret.

Regards,

Kevin Long

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Re: [asterisk-users] what to do when a sip password includes a semicolon

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/11/2016 01:43:47 PM:

> From: Saint Michael 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> , 
> Date: 03/11/2016 01:44 PM
> Subject: [asterisk-users] what to do when a sip password includes a 
semicolon
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> ​I got a new sip account, and the format
> register=> user:passwrd@proxy:port
> fails when the sip password ​has a semicolon
> Is there a possible workaround?
> I cannot change the password, it comes from the provider.

Try escaping the semicolon with a backslash. A password of abc;123 would 
become abc\;123
Not entirely certain that would work, but it would be the first thing I 
would try.

Also, I think a provider would be amenable to changing a password if it 
was problematic for some reason, but try the backslash first.

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Re: [asterisk-users] 2 devices same *actual* extension - can it be done

[Kevin Larsen]

> Can someone tell me if this is possible?
> 
> I currently have a VOIP phone registered on an Asterisk PBX at a 
> remote location (working fine).
> I want to install an Asterisk PBX at the local location. I will be 
> porting the current POSTS lines to SIP trunking.
> So now I want the remote line and the local lines to appear on the 
> same handset.
> This would mean I would have to pass internet to the phone for the 
> remote extension and also register the local extensions.
> So, for example, I could have the remote extension assigned to line 
> one (ACCOUNT 1 on the Polycom handset), and the local extensions 
> assigned to lines two, three, and four ( ACCOUNTS 2,3,4).
> 
> How do I do this?
> 

So, the first thing you will have to do is to make sure that your phone 
has routes to and can talk to each pbx over the network. Depending on your 
network design, this may be pretty simple or it may get pretty complex and 
will be hard to give a definitive answer in this discussion without more 
details. A good test might be to see if the phone can ping the pbx. Since 
you specifically mentioned a Polycom handset, look under 
Menu-Status-Diagnostics-Network-Ping. This will possibly help you to know 
that you can reach the pbx from the phone (provided your network is set up 
correctly and the pbx responds to pings). Note, many network designs will 
actually block pings even when the SIP and RTP traffic will traverse it 
just fine, so a failure here isn't necessarily the kiss of death.

Next, you will need to set up your phone to register with each PBX. 
Polycom has excellent docs on how to perform a setup using xml 
configuration files. Here is an example with four lines connecting to four 
different voip servers on a Polycom phone. Please note that I do not 
endorse the insecure usernames and passwords used here. They don't follow 
best practices and are only here for an example.



Note that this is just one small section out of a much larger 
configuration file used to completely configure a Polycom phone. Assuming 
you have the rest of your configs working, this would then put 4 lines 
onto the phone, each pointing to a different pbx and each labeled 
uniquely.

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Re: [asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

[Kevin Long]

Thanks John,


For anyone reading this using FreePBX - simply switching the default conference 
app from MeetMe to ConfBridge seems to be a drastic improvement, have not 
stress tested but running a conf now with no stutter on Confbrdige app.

Cheers,

Kevin Long



> On Mar 9, 2016, at 12:17 PM, Tech Support <aster...@voipbusiness.us> wrote:
> 
> One of the things you can do is google "app_konference". It doesn't require
> a clock source and is a very good application. I've successfully been using
> it for years and have had no problem with 100+ users in a single conference.
> Regards;
> John V.  
> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Long
> Sent: Wednesday, March 09, 2016 2:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] conference call stuttering / clocking issue (?) -
> ESXi virtual environment
> 
> 
> 
> Title says it all - for the time being I am stuck deploying Asterisk in ESXi
> . We are also looking at Proxmox for our next round of servers.. 
> 
> Everything works fine except conference calls - very stuttery , have tried a
> few different codecs.  I assume this is a granular clocking issue , and
> wondering if anyone has anything I could try to fix or mitigate the problem
> in ESXi environment .
> 
> We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8
> pjsip .
> 
> Thank you again,
> 
> 
> Kevin Long
> 
> 
> 
> 
> 
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[asterisk-users] conference call stuttering / clocking issue (?) - ESXi virtual environment

[Kevin Long]

Title says it all - for the time being I am stuck deploying Asterisk in ESXi . 
We are also looking at Proxmox for our next round of servers.. 

Everything works fine except conference calls - very stuttery , have tried a 
few different codecs.  I assume this is a granular clocking issue , and 
wondering if anyone has anything I could try to fix or mitigate the problem in 
ESXi environment .

We have freepbx (asterisk 11 chan_sip) and test environments asterisk 13.7/8 
pjsip .

Thank you again,


Kevin Long





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[asterisk-users] 2 devices same *actual* extension - can it be done

[Kevin Long]

Hello,

My company has invested heavily in Counterpath’s Stretto provisioning platform 
for Mobile and Desktop VoIP clients .

At this time their system allows 2 devices (for example iPhone + desktop 
computer) using the same software license per user , which many of our users 
require.

Their provisioning system assumes that both devices will use the same SIP 
extension for auth however. 


Normally we would use separate extensions and a follow-me , but if there is any 
way to use the same extension,  I need to figure it out. 

Thank you,

Kevin Long

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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

I can’t quite figure it out , I went ahead and pulled everything yet again, and 
I made sure to delete everything related to pjproject from my system, all the 
PJ  lib and include files that were in /usr/lib/  ,  I pulled pjproject from 
svn , pulled asterisk code from gerrit, recompiled everything, but still I 
think new TLS transports are being made which fail in my NAT scenarios .  I 
check with: 

tcpdump -i any src host 10.50.55.10  and  'tcp[13] & 2 != 0’ 


I see tcpdump print a new tcp SYN packet when I  try to make a call between 
endpoints and also when Asterisk tries to send OPTIONS command to the endpoint .

From my endpoints, I can call the “echo” applications and the call works fine, 
but I cannot call from one endpoint to another endpoint , even though they are 
both egistered. It does not say “unavailable’ or anything,  I see in the pjsip 
log that an INVITE is  “sent” , but I think the logger is just showing me that 
the INVITE message has been created, but it never reaches the endpoint because 
of the new TLS connection failing because of the NAT. Eventually, the call 
times out with a 408 error in the pjsip log.

I also see some log entries:
[Mar  4 12:29:10] DEBUG[16225] pjsip:   tlsc0x7f311400 TLS connect() error: 
Connection timed out [code=120110]
[Mar  4 12:29:29] DEBUG[16225] pjsip:   tlsc0x7f311400 TLS connect() error: 
Connection timed out [code=120110]




Just to be clear I am getting pjproject like so : 
svn co http://svn.pjsip.org/repos/pjproject/trunk


and asterisk :
git clone -b 13 http://gerrit.asterisk.org/asterisk



then I go to pjproject directory,  create a site_config.h file (to increase TLS 
connectors and set other options recommended on Wiki)

configure pjproject with the following options:

./configure --prefix=/usr --enable-shared --disable-sound --disable-resample 
--disable-video --disable-opencore-amr --with-external-srtp



Then go to asterisk directory

make clean; make distclean; ./boostrap.sh ; ./configure;  make menuselect; 
make; make install;










> On Mar 4, 2016, at 7:33 AM, George Joseph <george.jos...@fairview5.com> wrote:
> 
> 
> 
> On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.l...@haloprivacy.com> wrote:
> Hi George the patch was from here , you wrote it I believe . I pulled 
> asterisk 13 from git, apply this patch which fixed RTP issue , but I think 
> tla transport issue came back for me . 
> 
> https://gerrit.asterisk.org/#/c/2346/
> 
> ​Oh, that one, OK.  ​  It should be merged now so if you 'git pull' on 13 
> now, you should get it.  The transport re-use issue was in pjproject so is it 
> possible that you're not compiling against the latest trunk?
> 
> 
> 
> 
>  
> 
> Thank you
> 
> Sent from my iPhone
> 
> On Mar 4, 2016, at 12:01 AM, George Joseph <george.jos...@fairview5.com> 
> wrote:
> 
>> 
>> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.l...@haloprivacy.com> 
>> wrote:
>> 
>> Thanks George I appreciate the info .  Being able to see what codec is in 
>> use for call in progress is very handy sometimes.
>> 
>> As far as the RTP stats goes,  I see there is some info with “rtp” and 
>> “rtcp” commands which can be useful for troubleshooting. A running tally of 
>> # packets or bandwidth used would be awesome in along with the codec in 
>> "pjsip show channels" or something like that.
>> 
>> 
>> Im not certain, but I think the TLS signalling problem from this email may 
>> be happening to me again after patching for another pjsip/NAT issue which 
>> was with the external_media_address not working and the internal IP being 
>> sent in the SDP from asterisk - I applied this patch to the codebase and 
>> recompiled I am seeing the TLS “new transport”  issue again , I think.
>> 
>> ​I've lost track of who's applying what patches to ​which codebase. :)
>> 
>> Which patch did you apply for "external_media_address not working"?
>> 
>>  
>> 
>> Regards,
>> 
>> Kevin Long
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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

Hi George the patch was from here , you wrote it I believe . I pulled asterisk 
13 from git, apply this patch which fixed RTP issue , but I think tla transport 
issue came back for me . 

https://gerrit.asterisk.org/#/c/2346/

Thank you

Sent from my iPhone

> On Mar 4, 2016, at 12:01 AM, George Joseph <george.jos...@fairview5.com> 
> wrote:
> 
> 
> 
>> On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.l...@haloprivacy.com> 
>> wrote:
>> 
>> Thanks George I appreciate the info .  Being able to see what codec is in 
>> use for call in progress is very handy sometimes.
>> 
>> As far as the RTP stats goes,  I see there is some info with “rtp” and 
>> “rtcp” commands which can be useful for troubleshooting. A running tally of 
>> # packets or bandwidth used would be awesome in along with the codec in 
>> "pjsip show channels" or something like that.
>> 
>> 
>> Im not certain, but I think the TLS signalling problem from this email may 
>> be happening to me again after patching for another pjsip/NAT issue which 
>> was with the external_media_address not working and the internal IP being 
>> sent in the SDP from asterisk - I applied this patch to the codebase and 
>> recompiled I am seeing the TLS “new transport”  issue again , I think.
> 
> ​I've lost track of who's applying what patches to ​which codebase. :)
> 
> Which patch did you apply for "external_media_address not working"?
> 
>  
>> 
>> Regards,
>> 
>> Kevin Long
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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

Thanks George I appreciate the info .  Being able to see what codec is in use 
for call in progress is very handy sometimes. 

As far as the RTP stats goes,  I see there is some info with “rtp” and “rtcp” 
commands which can be useful for troubleshooting. A running tally of # packets 
or bandwidth used would be awesome in along with the codec in "pjsip show 
channels" or something like that.


Im not certain, but I think the TLS signalling problem from this email may be 
happening to me again after patching for another pjsip/NAT issue which was with 
the external_media_address not working and the internal IP being sent in the 
SDP from asterisk - I applied this patch to the codebase and recompiled I am 
seeing the TLS “new transport”  issue again , I think.

Regards,

Kevin Long

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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

So the patch did resolve the audio RTP issue and I can make echo calls now,   
but it seems like the last issue I posted to the list,  (pjsip driver making 
new outbound TLS transports instead of using existing SIP connection, not NAT 
friendly)   is happening again ..   Could that be?


Thanks again,


Kevin Long

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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

Hi Joshua,

This Asterisk 13 was pulled from git master branch just 2-3 days ago: 
GIT-13-d1495b . 

I used this very recent source code to overcome a pjsip problem (you can see my 
email list post from a few days ago)


Thanks again




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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

Hi Joshua,



Looking at the transmitted SIP packets from Asterisk,  it looks like Asterisk 
is only sending it’s own internal IP (it is behind a NAT too, with proper port 
forwarding) .

I did set in my transport the external_signaling_address and 
external_media_address  ,  and I have now put transport= into my endpoint 
configuration hoping they will “inherit” the correct public IP for the media .

But Asterisk is still sending RTP to the wrong IP .  


I am trying to test a “real world” scenario of public IP and NAT traversal,  
but I do have split tunnel VPN in my environment so the endpoint and the 
asterisk server *could* reach each other by the private IP ,but I am actually 
trying to avoid this with a proper configuration since my real users will not 
be on any  VPN, mostly. 







;===TRANSPORT
 




[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
local_net=10.50.55.0/24
external_media_address=66.114.139.174
external_signaling_address=66.114.139.174
cert_file=/etc/asterisk/keys/dev1.crt
priv_key_file=/etc/asterisk/keys/dev1.key
ca_list_file=/etc/asterisk/keys/ca.crt
cipher=AES256-SHA
method=tlsv1
 
;===EXTENSION 6001
 
[6000]
type=endpoint
context=internal
disallow=all
allow=ulaw
transport=transport-tls
auth=auth6000
aors=6000
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes


[auth6000]
type=auth
auth_type=userpass
password=6000
username=6000
 
[6000]
type=aor
qualify_frequency=30
max_contacts=1
remove_existing=yes


;===EXTENSION 6001

[6001]
type=endpoint
context=internal
disallow=all   
allow=ulaw
transport=transport-tls
auth=auth6001
aors=6001
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes



[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
qualify_frequency=30
max_contacts=1
remove_existing=yes










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Re: [asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

Thank you for the response Joshua . 


I had rtp_symmetric=yes  before I wrote the email,   then I set it to no,  
restart asterisk, and tried to make the call from the remote endpoint again but 
still tcpdump is showing me the RTP packets are being sent from Asterisk to the 
private IP.

tcpdump on asterisk server showing UDP packet bound for my remote endpoints 
internal IP:
17:07:57.130212 IP 10.50.55.10.6214 > 10.128.30.239.51126: UDP, length 182




Current pjsip.conf file 


[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
local_net=10.50.55.0/24
external_media_address=
external_signaling_address=
cert_file=/etc/asterisk/keys/dev1.crt
priv_key_file=/etc/asterisk/keys/dev1.key
ca_list_file=/etc/asterisk/keys/ca.crt
cipher=AES256-SHA
method=tlsv1
 
;===EXTENSION 6001
 
[6000]
type=endpoint
context=internal
disallow=all
allow=ulaw
auth=auth6000
aors=6000
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=no
media_encryption=sdes


[auth6000]
type=auth
auth_type=userpass
password=6000
username=6000
 
[6000]
type=aor
qualify_frequency=30
max_contacts=1
remove_existing=yes


;===EXTENSION 6001

[6001]
type=endpoint
context=internal
disallow=all   
allow=ulaw
auth=auth6001
aors=6001
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=no
media_encryption=sdes



[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
qualify_frequency=30
max_contacts=1
remove_existing=yes

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[asterisk-users] RTP / NAT question ( pjsip )

[Kevin Long]

I am having trouble with RTP and NAT :


Below is a SIP SDP invite from a remote endpoint which is trying to call 
extension 420 which is the ECHO application .


As you can see, the public IP is where the request comes in from,  but the SDP 
contains the private, internal IP in numerous places.


I do have rewrite_contact=yes;  on in my pjsip endpoint configuration,  but 
still the “rtp set debug on” command is showing me that when I dial into the 
echo application,  RTP packets are being sent to the private IP and not the 
public IP .



Advice appreciated thank you. 



<--- Received SIP request (1282 bytes) from TLS:72.52.31.109:55256 --->
INVITE sip:4...@dev1.domain.com SIP/2.0
Via: SIP/2.0/TLS 
10.128.30.239:55253;branch=z9hG4bK-524287-1---bf28eb29eb900b43;rport
Max-Forwards: 70
Contact: <sip:6000@72.52.31.109:55256;transport=TLS;rinstance=e652ef90f2843e40>
To: <sip:4...@dev1.domain.com>
From: "Kevin"<sip:6...@dev1.domain.com>;tag=0af40611
Call-ID: MGE5OWFhMDY5OGFhYzM4ZDIxNjA5OGRjY2M5OWE3ZGY
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, INFO, NOTIFY, UPDATE, PRACK, MESSAGE, 
OPTIONS, SUBSCRIBE, OPTIONS
Content-Type: application/sdp
Supported: replaces, 100rel
User-Agent: Bria iOS release 3.6.2 stamp 33024
Authorization: Digest 
username="6000",realm="asterisk",nonce="1456965577/29f2977e5352209d33847b1eafc5f937",uri="sip:4...@dev1.haloprivacy.com",response="9c23bba47f43fa343bfc3bd2580a84ad",cnonce="ea996236e91c869bb16b1652c8504ba3",nc=0001,qop=auth,algorithm=md5,opaque="609ab4014ccfac10"
Content-Length: 358

v=0
o=- 1456965576139402 1 IN IP4 10.128.30.239
s=Cpc session
c=IN IP4 10.128.30.239
t=0 0
m=audio 61216 RTP/SAVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 
inline:tkUxPSw8qTZ25fk6VuQPWNVOABk5mwe63/+d7vP7
a=crypto:2 AES_CM_128_HMAC_SHA1_32 
inline:tkUxPSw8qTZ25fk6VuQPWNVOABk5mwe63/+d7vP7
a=sendrecv



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Re: [asterisk-users] PJSIP signaling question

[Kevin Long]

Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject 
from the SVN and will test accordingly .



I have a few more questions about PJSIP in Asterisk 13:


1.  Is there any way to list current ongoing calls and see what codecs are 
being used in the RTP streams?  With chan_sip,  “sip show channels” did this.  

2. Also with a PJSIP initiated call, is there a way to see how man RTP packets 
have been sent and received for the call , I am debugging some intermittent 
1-way and no-way audio on calls , and I am having trouble figuring out fi it is 
the client, firewall, or Asterisk/pjsip that is the culprit .


Regards,

Kevin Long

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[asterisk-users] PJSIP signaling question

[Kevin Long]

Greetings.


I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile 
apps operating in environments that I do not control. 

 I would like Asterisk to default to sending INVITES and all other SIP signals 
to endpoints via the existing SIP TLS connection which is already established, 
rather than trying to create a new TLS connection to an endpoint which is 
likely behind a NAT which will not allow a new inbound TCP/TLS connection.


My experience with chan_sip suggest to me that this was the default behavior, 
or more likely a fallback behavior, because I never had this issue before with 
endpoints not receiving INVITES so long as they were registered and had an open 
SIP control connection.


I thought that I could avoid these failed outbound connections by commenting 
out the “transport” option on my endpoint configurations, but tcpdump is 
showing me that asterisk is still trying to create *new* TLS outbound 
connections to my endpoints, which are failing.




Thank you for your time

Kevin


-




My simple pjsip config file:





[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
local_net=10.50.55.0/24
external_media_address=x.x.x.x
external_signaling_address=x.x.x.x
cert_file=/etc/asterisk/keys/dev1.crt
priv_key_file=/etc/asterisk/keys/dev1.key
ca_list_file=/etc/asterisk/keys/ca.crt
cipher=AES256-SHA
method=tlsv1
 
;===EXTENSION 6001
 
[6000]
type=endpoint
context=internal
disallow=all
allow=ulaw
;transport=transport-tls
auth=auth6000
aors=6000
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes


[auth6000]
type=auth
auth_type=userpass
password=6000
username=6000
 
[6000]
type=aor
max_contacts=1
remove_existing=yes


;===EXTENSION 6001

[6001]
type=endpoint
context=internal
disallow=all   
allow=ulaw
;transport=transport-tls
auth=auth6001
aors=6001
direct_media=no
rewrite_contact=yes  ; necessary if endpoint does not know/register public 
ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes



[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
max_contacts=1
remove_existing=yes

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Re: [asterisk-users] Passing Caller ID through Digium Gateway

[Kevin Larsen]

> Hi All,
> 
> I've setup a Digium G100 VoIP gateway to replace an internal PCI VoIP 
> card in our Asterisk PBX.  When using the VoIP card the callerid entries 

> listed in sip.conf were displayed when calling someone over the PSTN. 
> Now, however, though the gateway it just displays the default number 
> assigned to our PRI.  I'm wondering if anyone having experience with the 

> Digium gateways can point me in the right direction to have the gateway 
> respect the callerid entries listed in sip.conf.  We are using an older 
> Asterisk 1.6 build.

We use G100 and 200s at a few of our sites and caller id passes through 
just fine. Check under Configuration -> call routing rules to make sure 
you don't have a caller ID name and number set in there. You have to take 
it off of Simple Entry Mode to see the options.

Also, on your SIP Endopoint configuration, under the call settings tab, 
make sure you have your Caller ID Presentation set correctly. From the 
help on that option:

Caller ID Presentation:Handles Caller ID presentation on outgoing calls. 
Allow for prohibiting Caller ID presentation, and defines whether the 
information has been screened by an authoritative source. Options other 
than screening, allowed, and prohibited indicate that the Caller ID was 
provided by the network. 

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[asterisk-users] Determining and setting TLS cipher ?

[Kevin Long]

Greetings,


I use TLS transport for all my endpoints on my production system (Asterisk 11) 
.  I need to debug some NAT traversal issues, and would like to use the 
‘sngrep’ tool which shows SIP messages from a packet capture.  Per the 
developer of ‘sngrep’ : "Right now, sngrep only supports 
TLS_RSA_WITH_AES_128_CBC_SHA and TLS_RSA_WITH_AES_256_CBC_SHA”


I have not specified a cipher for my sip.conf TLS transport and I do not know 
how to see which one is being used .


The list of ciphers I see available to me, at least based on running “openssl 
ciphers” command on my Asterisk box, are listed below. None of them exactly 
matches the strings above listed as supported ciphers for sngrep.  


Can I configure Asterisk to use one of the ciphers supported by sngrep? Is 
there a better tool than sngrep for viewing TLS SIP captures?  Are the sngrep 
supported ciphers safe?


Thank you,

Kevin Long


output from “openssl ciphers” on my Asterisk box:

ECDHE-RSA-AES256-GCM-SHA384
ECDHE-ECDSA-AES256-GCM-SHA384
ECDHE-RSA-AES256-SHA384
ECDHE-ECDSA-AES256-SHA384
ECDHE-RSA-AES256-SHA
ECDHE-ECDSA-AES256-SHA
DHE-DSS-AES256-GCM-SHA384
DHE-RSA-AES256-GCM-SHA384
DHE-RSA-AES256-SHA256
DHE-DSS-AES256-SHA256
DHE-RSA-AES256-SHA
DHE-DSS-AES256-SHA
DHE-RSA-CAMELLIA256-SHA
DHE-DSS-CAMELLIA256-SHA
ECDH-RSA-AES256-GCM-SHA384
ECDH-ECDSA-AES256-GCM-SHA384
ECDH-RSA-AES256-SHA384
ECDH-ECDSA-AES256-SHA384
ECDH-RSA-AES256-SHA
ECDH-ECDSA-AES256-SHA
AES256-GCM-SHA384
AES256-SHA256
AES256-SHA
CAMELLIA256-SHA
PSK-AES256-CBC-SHA
ECDHE-RSA-AES128-GCM-SHA256
ECDHE-ECDSA-AES128-GCM-SHA256
ECDHE-RSA-AES128-SHA256
ECDHE-ECDSA-AES128-SHA256
ECDHE-RSA-AES128-SHA
ECDHE-ECDSA-AES128-SHA
DHE-DSS-AES128-GCM-SHA256
DHE-RSA-AES128-GCM-SHA256
DHE-RSA-AES128-SHA256
DHE-DSS-AES128-SHA256
DHE-RSA-AES128-SHA
DHE-DSS-AES128-SHA
ECDHE-RSA-DES-CBC3-SHA
ECDHE-ECDSA-DES-CBC3-SHA
DHE-RSA-SEED-SHA
DHE-DSS-SEED-SHA
DHE-RSA-CAMELLIA128-SHA
DHE-DSS-CAMELLIA128-SHA
EDH-RSA-DES-CBC3-SHA
EDH-DSS-DES-CBC3-SHA
ECDH-RSA-AES128-GCM-SHA256
ECDH-ECDSA-AES128-GCM-SHA256
ECDH-RSA-AES128-SHA256
ECDH-ECDSA-AES128-SHA256
ECDH-RSA-AES128-SHA
ECDH-ECDSA-AES128-SHA
ECDH-RSA-DES-CBC3-SHA
ECDH-ECDSA-DES-CBC3-SHA
AES128-GCM-SHA256
AES128-SHA256
AES128-SHA
SEED-SHA
CAMELLIA128-SHA
DES-CBC3-SHA
IDEA-CBC-SHA
PSK-AES128-CBC-SHA
PSK-3DES-EDE-CBC-SHA
KRB5-IDEA-CBC-SHA
KRB5-DES-CBC3-SHA
KRB5-IDEA-CBC-MD5
KRB5-DES-CBC3-MD5
ECDHE-RSA-RC4-SHA
ECDHE-ECDSA-RC4-SHA
ECDH-RSA-RC4-SHA
ECDH-ECDSA-RC4-SHA
RC4-SHA
RC4-MD5
PSK-RC4-SHA
KRB5-RC4-SHA
KRB5-RC4-MD5




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[asterisk-users] NAT traversal for mobile app softphones - best strategy?

[Kevin Long]

Greetings,


My asterisk systems sit behind a Meraki mx80 firewall at a data center.  I use 
static public IPs on the firewall and port forward  5060,5061, and 
10,000-20,000 so the clients can connect. Per Meraki support: "Our MX security 
appliances do not support SIP ALG.  Our NAT is a stateful NAT, so only return 
traffic will be able to traverse the NAT, unless a port forwarding rule is in 
place.” Im not sure if this would have any negative impact or if my traversal 
issues are only client side.  My port forwarding should be good I think.

Especially since testing with asterisk 13.7 and PJSIP (compared with freepbx 
chan_sip asterisk 11)  I am having more problems with 1-way and no-way audio .

Most of my endpoints are iPhones using the “Bria” soft phone app from 
Counterpath. This means that their IP address may change often, and whatever 
kind of NAT they are behind is beyond my control. 

Given this scenario, I’m hoping for advice on the best strategy for 
configuration of my Asterisk server, and soft phones with ICE/TURN/STUN?  To 
help with NAT traversal. The Bria app allows multiple options to be turned on 
for traversal strategy:


For SIP:
RPORT WiFi
RPOR TMobile
Outbound Wifi
Outbound Mobil
STUN WiFi
STUN Mobile

-
STUN/TURN  (server/username/password fields)
-
Media NAT Traversal
STUN WiFi
Stun Mobile
Use ICE Wifi
Use ICE Mobile
Use TURN WiFi
Use TURN Mobile



—


To use ICE on Asterisk, do I need to also set up a separate TURN server, and is 
one in particular recommended? I’ve looked into "turnserver" and 
"resiprocate-turn-server" (reTurn) briefly. I’m unclear as to whether I need to 
run this server on a true public IP or if the server can also run behind a 
firewall with port forward from the WAN public IP.  I’m also unclear as to 
whether I truly need 2 separate public IPs for the turn server to work, which I 
have seen mentioned in some of the documents.


Thank you for your time.

Regards,

Kevin Long





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Re: [asterisk-users] Queue logfile txt format in mySQL needed

[Kevin Larsen]

> From: Thomas 
> To: asterisk-users@lists.digium.com, 
> Date: 01/21/2016 04:17 AM
> Subject: [asterisk-users] Queue logfile txt format in mySQL needed
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Hello,
> 
> Iam using queues and agents, thats OK.
> 
> I have interesting information form Asterisk in txt file format
> var/log/asterisk/queue_log
> 
> Today Iam reading these txt files and wrote them in an mySQL databases.
> 
> I would need this information more realtime. Some information I do 
writing in 
> the dialplan direct in an mySQL database.
> 
> Is there any way that Asterisk write this information direct in an mySQL 

> database instead of using var/log/asterisk/queue_log?

I haven't done this myself, but it looks like you just need to set up the 
appropriate database connections. See here for a semi-recent example:
http://stackoverflow.com/questions/30161384/asterisk-11-queue-log-to-mysql

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Re: [asterisk-users] Statsd Dialplan Application

[Kevin Harwell]

On Tue, Jan 19, 2016 at 8:46 AM, Annus Fictus <annusfic...@gmail.com> wrote:

> Hello,
>
> I'd like to do some tests with the StatsD dialplan application but on the
> last version of Asterisk 13 (13.7.0) I can't find this application.
>
> New Features made in this release:
> ---
>  * ASTERISK-25419 - Dialplan Application for Integration of StatsD
>   (Reported by Ashley Sanders)
>
> res_statsd module are correctly compiled y loaded.
>
> Any hint?
>
> Regards
>
>
>

Unfortunately these changes did not go out with the latest release of
13.7.0. Actually the new StatsD Dialplan application currently resides in
master only. A small change to the res_statsd api was made and got tagged
with that issue number for some reason, thus making it look as if the
StatsD application feature was added to 13.


-- 

Kevin Harwell
Digium, Inc. | Software Developer
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] Forwarding call if extension busy

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/04/2016 08:55:40 AM:

> My question:
> 
> - two extensions:  and 
> - an active call on 
> - incoming calls to  should be forwarded to  (call advice!) and 

> 
> I know how can I forward an incoming call to more than an extension, 
> but I have no idea how can I get the information, that  has 
> already an active call...
> 

I am not sure if I completely understand what you are trying to do, but it 
sounds like you want to query the DEVICE_STATE function.

For instance, my customer service department has this thing against ever 
having their phone ring a call while they are already on a call, so for 
these special little snowflakes, I have  the following line:

same => n(voice),GotoIf($["${DEVICE_STATE(sip/${EXTEN})}" != 
"NOT_INUSE"]?voicebusy)

Basically, this little line looks at the extension and if it shows 
anything other than free (NOT_INUSE), it jumps to the voicebusy line in 
the dialplan. The voicebusy line just hits voicemail directly.

You can use this same idea to branch your logic and handle a variety of 
situations. In my case, I only want to actually perform the dial if the 
phone is currently not in use, so my logic was fairly simple.

See here for reference:
https://wiki.asterisk.org/wiki/display/AST/Device+State

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Re: [asterisk-users] Forwarding call if extension busy

[Kevin Larsen]

> Kevin Larsen <kevin.lar...@pioneerballoon.com> schrieb:
> 
> > I am not sure if I completely understand what you are trying to do, 
but it 
> > sounds like you want to query the DEVICE_STATE function.
> 
> IT WORKS
> 
> Thank you very much!
> 

Glad I was able to help. You are most welcome.

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[asterisk-users] How exactly does asterisk know what IP to send RTP traffic to?

[Kevin Long]

Hello,

I have a somewhat confusing use case.  We use a mobile voip app and our users 
connect to our PBX via a public IP of our firewall which port forwards to 
asterisk (TLS and SRTP ports). Works fine.

Sometimes however, our users are also connected to our VPN (LT2P/Ipsec) which 
is served by the same firewall that our PBX sits behind at the datacenter.

In this case, most often the calls go through but there is no audio.  

I believe that asterisk “thinks” in this case that the IP of the clients,  to 
send RTP traffic to ,t is the firewall’s IP, rather than the IP that the VPN 
server assigned the client device. 

Does asterisk send RTP traffic to the IP which is in the IP headers of the SIP 
REGISTER , or can a client “specify” it’s truly reachable IP ?

I hope this makes sense. 

Regards,

Kevin Long




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[asterisk-users] repeating TLS error in log file

[Kevin Long]

Greetings,


I use TLS and SRTP on all my extensions.  I use openssl and distribute my root 
certificate to my endpoints.   Most of the time my calls work just fine.  

Sometimes I receive a repeating error in my log files however, and I don’t know 
why this is happening. I’m wondering if this is really from the TLS connection 
for SIP, or an underlying error with SRTP decoding..   I sometimes get this 
message in the log when things seems to be working fine.  Is there a better way 
to debug exactly why I’m getting this error? Sometimes I have dozens of these 
errors in a row.

My openssl certificate chain checks out fine  with openssl verify command ..




[2015-10-26 12:23:42] WARNING[9915] tcptls.c: FILE * open failed!
[2015-10-26 12:23:42] VERBOSE[9916] tcptls.c:   == Problem setting up ssl 
connection: error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca

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[asterisk-users] AUTO: Kevin Larsen is out of the office (returning 10/24/2015)

[Kevin Larsen]

I am out of the office until 10/24/2015.

I am working in Mexico with limited availability. If the matter is urgent,
please contact the Pioneer Helpdesk.


Note: This is an automated response to your message  "Re: [asterisk-users]
Live Recording on the NAS?" sent on 10/15/2015 1:55:13 PM.

This is the only notification you will receive while this person is away.


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Re: [asterisk-users] Asterisk => Mediant 1000 (AudioCodes) => PSTN (E1)

[Kevin Larsen]

> 
> Does anyone have any information for me?
> 
> 
> Welinghton.
> 
> 
> 
> Citando Welinghton Magno Guimaraes :
> Hello!
> 
> I am setting up an Asterisk server with a Mediant 1000 (Audiocodes) 
> to make external links. Does anyone have any manual or instructions 
> on how to proceed?
> 
> Asterisk  =>  Mediant 1000 (AudioCodes)  =>  PSTN (E1)
> 
> 
> I will be very grateful for the help.
> 
> Thanks!
> 
> 
> Welinghton.
> 

If you do a search for mediant 1000 asterisk you will find some pages that 
might help. One of the problems I have found (I have a couple of 
AudioCodes devices), is that they do not publish anything resembling a 
useful manual to assist end users in setting up their devices. They want 
you to pay for a support contract and for install services instead.

I figured mine out by a lot of trial and error, unfortunately. My devices 
were for fxs/fxo, so unfortunately I doubt me experience would be much 
help.
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Re: [asterisk-users] Share agents state?

[Kevin Larsen]

> Is it possible to share all agents state? if an agent is on the 
> phone on a queue on one of the Asterisk servers, other servers will 
> need to about it and therefore, will be able to operate adequately?
> For instance, an agent is a member of two queues (app_queue 
> realtime) and those queues on separate server.
> Thanks 

You can indeed share a phone's state between servers. If using chan_sip, 
you will be looking at doing something like XMPP. If you are doing pjsip, 
you can do it directly without needing the xmpp server. 

For pjsip, look at 
https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP

For chan_sip, look at 
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub

For the record, I have done the xmpp setup and it works well, but there 
was a pretty steep learning curve involved in getting everything working. 
I haven't had a chance to look at upgrading to Asterisk 13 and pjsip to 
set it up, but the configuration looks to be much easier.

I use it because I have Site A which hosts a customer service call queue 
where most of the agents exist on the Site A server. However, I have two 
agents who are at Site B and we don't want to send them a call from the 
Site A queue if they are already on a call from the Site B server. Seems 
to work and no complaints from the group that uses it.-- 
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Re: [asterisk-users] How to integrate Asterisk with XMPP

[Kevin Larsen]

> 
> How to integrate Asterisk with XMPP ?
> 

What you are asking for isn't a simple question to answer. What exactly do 
you want to accomplish by integrating XMPP? Shared states among multiple 
extensions? Passing messages between extensions? Depending on what you 
want and what infrastructure you have in place will all influence the 
answer.

Also, you will get better responses if you say what you have tried and 
what isn't working or say what you goal is and ask for pointer on how to 
get there. Depending on what you want to do, there are multiple tutorials 
available online, but I will say that I did find it was a bit of trial and 
error to get xmpp working in my organization. I use it for allowing 
extensions on remote sites to join in to some of our call queues, thus 
needing our (multiple) asterisk boxes to be able to share extension states 
with each other. It wasn't the easiest thing in the world to get working 
on the 11 series. 

Depending on what you want to do, the new pubsub features in PJSIP in 
Asterisk 13 series may do what you want. I know I am looking forward to 
investigating them and quite possibly getting rid of my xmpp setup.

https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP-- 
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Re: [asterisk-users] Receiving faxes with spandsp question

[Kevin Larsen]

 I’m trying to add fax functionality to my asterisk installation. 
 Right now I’m focusing on receiving faxes. This is not explained in 
 a book, but I assume that I can use same context, add “fax” 
 extension and if someone calls to send fax - it will autodetect. Right?
 
  Per book, I made following setup additions:
 
 1. In sip.conf [general] I added:
 
 ;FAX stuff
 faxdetect=yes
 t38pt_udptl=yes
 
 2. In extensions.conf I hade something like this:
 
 [from-callcentric]
 exten = s,1,Goto(automated_attendant,s,1)
 
 ; FAX handling stuff AS IN BOOK
 exten = fax,1,Verbose(3,Incoming Fax)
 same = n,Set(FAXDEST=/tmp) ; folder where faxes will be 
stored
 same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
 same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
 same = n,Verbose(3, - Fax receipt completed with status: $
 {FAXSTATUS})
 
 Well, that didn’t work. Trying to send fax - it was going to my 
 autoattendant and never triggered fax. So, I made a change like so:
 
 3. Changed extensions.conf
 
 [from-callcentric]
 ; FAX handling stuff AS IN BOOK
 exten = s,1,Verbose(3,Incoming Fax)
 same = n,Set(FAXDEST=/tmp) ; folder where faxes will be 
stored
 same = n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)})
 same = n,ReceiveFax(${FAXDEST}/${tempfax}.tif)
 same = n,Verbose(3, - Fax receipt completed with status: $
 {FAXSTATUS})
 
 I just made it fax handling context, and I got FAX :)  But, 
 while fax was received I was getting following:
 
 [2015-06-24 23:40:28] WARNING[47369][C-0005]: res_fax_spandsp.c:
 438 spandsp_log: WARNING T.30 ECM carrier not found
 
 
 QUESTIONS:
 
 1. Should I do something about this warning?
 2. How do I receive fax and have main entry to auto attendant in a 
 same context? Can I have it on same puplic phone number?
 

I think your problem may be that even though you created the exten = fax 
line, it never has a chance to auto detect and go there as it has already 
left that context before it has detected the fax and then has no fax 
extension to redirect to. You could put your fax extension in the 
automated_attendant context and that should work. I recommend a slightly 
different way of handling faxes.

What I did was create an incoming fax context (fax_incoming). In your 
above example, that is where the fax extension would live. That way I can 
handle my reception of faxes in one spot and if I ever need to bug 
fix/change my dialplan, I only have to do it in one spot. Then anywhere 
that I want to autodetect faxes and move them to the fax context I put the 
following extension code:

exten = fax,1,Goto(fax_incoming,${dialednumber},1)

Of course, if you don't want the comment in there, that could be reduced 
to just one line. Also, ${dialednumber} is just a variable I use to hold 
the originally dialed number in case it has been altered as it goes 
through my dialplan so that I can have my CDR records show what was 
originally dialed in case I need to go back later. In your example, you 
would replace ${dialednumber} with whatever you need to work with your fax 
handler. I have multiple fax numbers, so I like to know which one was 
dialed to reach that spot. Makes bookkeeping easier.

I have this working on my sites that use an IVR and based on the timing, 
it gets a few seconds into playing the ivr message usually before it 
detects the fax and redirects it to the proper fax context. I have 
separate fax numbers, but this does catch those who don't pay attention 
and dial the main number instead of the fax number. I also use it on the 
direct dials to my phones. When I get a fax that way, my desk phone will 
get about one ring before the fax is detected and the call is moved away.

Faxing can be finicky to get working how you want it, but you can usually 
make it handle the faxes like you want.
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Re: [asterisk-users] asterisk email to fax

[Kevin Larsen]

 Since the O.P. said he's using it for his home office, I think he'll
 be able to control user expectations :-)


I provide tech support to my parents on all their computers. The amount of 
annoyance I have dealt with in the last few months over the fact that a 
recipe program and various card making programs designed for Windows 
3.1/95 won't run on my mom's Windows 7 64 bit computer tells me you are 
not as right as you think you are.-- 
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Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

[Kevin Larsen]

 The legal and medical communities still seem to prefer faxing, in 
 the ( mistaken? ) belief that it is more secure. In fact the medical
 community is fearful of the legal beagles.
 
 These groups are really slow to change.
 At least in the USA

The couple of times I have received medical faxes to my fax bank scare me 
about the actual security. My company is not in the medical field, nowhere 
close, in fact.

In one case, the fax included the patients name, address, phone, Date of 
Birth, SSN, and confidential medical history. The comment I made to a 
coworker was that if I wanted to steal an identity, they had just handed 
me everything I would need.

In the second case, it was a question from a pharmacy to a doctors office. 
Not quite so bad. I called up the pharmacy and said I had a problem with a 
fax they had sent. After asking me for some information from the fax so 
they could identify which patient I was calling about they asked what the 
problem was. I replied that I was a manufacturer of balloons and not a 
doctor's office. To say there was a bit of panic creeping into the guys 
voice on the other end was an understatement. I think I triggered some 
HIPAA reporting provisions.-- 
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Re: [asterisk-users] small homebrew pbx

[Kevin Larsen]

 I don't know this 'translates' to Italy, but this is what I would advise 

 somebody in the US to consider, assuming you have a reliable Internet 
 connection.
 
 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk 

 at Home.' A@H was an ancient distribution from around 2005.
 
 1) Rent a DID (a 'PSTN number') from a reputable SIP provider. This 
 eliminates the need for a PCI/USB interface and you won't disrupt your 
 'business' while you figure out how to configure and test your Asterisk 
 server.
 
 In the US, you can rent a DID for about $1.50 per month and about a 
$0.01 
 per minute of 'talk time.' For 10 calls per day, this should beat the 
hell 
 out of a 'landline' monthly standing fee.
 
 In the US, it costs less than $20.00 to 'port' your existing number if 
you 
 are really in love with it.
 
 2) Ditch the 'room warmer' and find something really small and cheap to 
 run. I live in San Diego and we pay $0.32 per kWh. I'd guess running 
your 
 rig would cost me $50.00 to $100.00 per month just in electricity -- and 

 probably that much again in the summer for additional Air Conditioning.
 
 Take a look at Soekris net4801. It's pretty old (but very reliable) and 
 it's CPU will limit you on what OS you can run, but it will give you an 
 idea of how small (and cheap to power) an 'Asterisk server' capable of 
 handling a couple of simultaneous calls can be.
 
 For a more modern server, look for something small and cheap based on 
 something like an Atom processor. Maybe a used laptop. If the battery is 

 still good, you've solved your UPS problem as well. Although, if you 
lose 
 power, you've probably lost your Internet connection as well so you 
could 
 only make calls between extensions.
 
 3) For the IP phones, check out ebay.com. Last year, I picked up 3 
Polycom 
 SP 501's for $20.00 each. A little dated, but a great phone.

I gotta agree with most all of this. Asterisk has been shown to run on a 
Raspberry Pi and the Raspberry Pi 2 and will handle a few simultaneous 
calls. Another resource is http://www.plugpbx.org/

For home use, I would think either would be a good low power way to run 
Asterisk. Unless you just really need the land line, ditch the analog line 
and go voip from start to finish.
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

 Very strange...
 I ran the Asterisk CLI for other tasks, and suddenly I got this message:
 
   == Using SIP RTP CoS mark 5
 -- Executing [000972592603325@default:1] Verbose(SIP/192.168.
 20.120-002a, 2,PROXY Call from 0123456 to 000972592603325) innew 
stack
   == PROXY Call from 0123456 to 000972592603325
 -- Executing [000972592603325@default:2] Set(SIP/192.168.20.
 120-002a, CHANNEL(musicclass)=default) in new stack
 -- Executing [000972592603325@default:3] GotoIf(SIP/192.168.20.
 120-002a, 0?dialluca) in new stack
 -- Executing [000972592603325@default:4] GotoIf(SIP/192.168.20.
 120-002a, 0?dialfax) in new stack
 -- Executing [000972592603325@default:5] GotoIf(SIP/192.168.20.
 120-002a, 0?dialanika) in new stack
 -- Executing [000972592603325@default:6] Dial(SIP/192.168.20.
 120-002a, SIP/pbxluca/000972592603325,,R) in new stack
 [Jun  8 21:42:50] WARNING[18981]: app_dial.c:2345 dial_exec_full: 
 Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [000972592603325@default:7] Hangup(SIP/192.168.20.
 120-002a, ) in new stack
   == Spawn extension (default, 000972592603325, 7) exited non-zero 
 on 'SIP/192.168.20.120-002a'
 [Jun  8 21:43:22] WARNING[16633]: chan_sip.c:3830 retrans_pkt: 
 Retransmission timeout reached on transmission 
 8dc31ca4e660a0408450715638784d86 for seqno 1 (Critical Response) -- See 
 https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
 Packet timed out after 32001ms with no response
 
 At the time no phone try to call...
 On my Firewall I see a SIP packet coming from an IP in Palestine...
 Am I cracked? I think I disabled all guest access. How can I check if 
my
 Asterisk allows guest to originate calls?

Based on SIP packets coming in from IP addresses you don't recognize, 
while you may not be hacked, you would seem to have people probing your 
system. One thing you can do at the firewall level is restrict inbound sip 
communications to only those from your external phone providers. Depending 
on their setup, they should be able to give you an IP, a range of IPs or a 
name that can be used (i.e. sip.myphoneprovider.com). If you restrict your 
inbound sip to that, it will be very helpful. Also, there are further 
steps you can take to harden your systems. An internet search will bring 
up many, but here are a couple of good ones:

http://blogs.digium.com/2009/03/28/sip-security/
http://www.ipcomms.net/blog/70-11-steps-to-secure-your-asterisk-ip-pbx
http://nerdvittles.com/?p=580-- 
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

 OK, I set alwaysauthreject = yes and I discovered a allowguest, which I 
set
 to no, too.
 The PBX is behind a Firewall and I just allow UDP 5060 and 1-10100.
 Now I log the SIP-pakets coming from Internet, too...
 
 Hopefully I solved my problem...

Make sure you have solved the problem. You don't want to get hit with a 
phone bill for calls from your location to Israel. Basically, they are 
hoping that you are running the equivalent of a mail server open relay. 
They are trying to use you to dial out to another number. You don't want 
to pay for these calls.

The calls are being dumped into your default context. It's not matching on 
your gotoif statements, so finally it is trying to execute this:
Dial(SIP/192.168.20.120-002a, SIP/pbxluca/000972592603325,,R) in 
new stack

Not sure what trunk pbxluca is, but if that is an outbound trunk, then 
this is very bad. The only reason it would fail then is if they have the 
outbound dial pattern wrong, which is a sure sign that you are open in the 
future to having someone make this kind of call in a way that does work 
and leaves you on the hook. Based on your email address, I am guessing you 
are in Germany. Looks like they almost have the correct outbound pattern 
for dialing from Germany to Israel. It should be 00972592603325 (notice 
the one less zero in the front). Please tell me that pbxluca is not an 
outbound dialing context? If it is, you need to fix this very quickly.-- 
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Re: [asterisk-users] Am I cracked?

[Kevin Larsen]

  Make sure you have solved the problem. You don't want to get hit with 
a 
  phone bill for calls from your location to Israel. Basically, they are 

  hoping that you are running the equivalent of a mail server open 
relay. 
  They are trying to use you to dial out to another number. You don't 
want 
  to pay for these calls.
 
 Of course, but how can I test, if I am an open relay?
 
  The calls are being dumped into your default context. It's not 
matching on 
  your gotoif statements, so finally it is trying to execute this:
  Dial(SIP/192.168.20.120-002a, SIP/pbxluca/000972592603325,,R) 
in 
  new stack
  
  Not sure what trunk pbxluca is, but if that is an outbound trunk, then 

  this is very bad. The only reason it would fail then is if they have 
the 
 
 This is one of my outbound trunk...
 
  outbound dial pattern wrong, which is a sure sign that you are open in 
the 
  future to having someone make this kind of call in a way that does 
work 
  and leaves you on the hook. Based on your email address, I am guessing 
you 
  are in Germany. Looks like they almost have the correct outbound 
pattern 
  for dialing from Germany to Israel. It should be 00972592603325 
(notice 
  the one less zero in the front). Please tell me that pbxluca is not an 

  outbound dialing context? If it is, you need to fix this very quickly.
 
 How can I fix it? Of course, I need to be able to call any phone on this
 world...
 On a Mail-Server I'd restrict outgoing calls to authenticated users. I 
was
 sure, that Asterisk already do that, but I'm not sure anymore...
 How can I restrict it?

I am sure others can chime in, but first things first, you want inbound 
calls and outbound calls to be in different contexts. Don't let your 
default context reach an outbound line. Your registered phones will be in 
a context that can call out which should be different from the default.

Also, make sure that your phones are registering with passwords (secret) 
that are different than the extension number. Makes it harder to guess.

The big thing to keep in mind dialplan wise is to never let an inbound 
call have a path to loop back outbound. The two of the biggest vectors for 
fraud will be allowing a non-authenticated sip call to get outbound over 
your trunks and to have weak credentials that can be cracked that will let 
someone else impersonate your phones.

And you can still wipe out most fraud by restricting the IP addresses you 
let in from the outside world. I prefer to have the most restrictive 
communications I can and then fix it if I discover that something doesn't 
work. Better to fail and fix than to permit and pay for it later. The 
providers I tend to like best not only give me what I need to restrict to 
their IP ranges, but also put in place restrictions on their end to only 
talk to my account from my external static IP address. That way someone 
could figure out my credentials, but if they can't spoof my ip address it 
still won't work. That is dependent on what the provider can do though.-- 
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Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 I love this question, simply because it allows me to talk about one 
 of the neatest features I programmed into my system that barely 
 anyone knows exists. Plus it lines up pretty much exactly with what 
 you are trying to do. 
 
 We have our gate control system tied into our Asterisk phone system 
 so it is possible to dial a code on the phone and open the entrance 
 gate to let someone in after hours. Only problem is this happens so 
 rarely that no one (myself included) ever remembered the code. Thus 
 a search for a better way. 
 
 Now, when someone uses the gate phone to request entry, I change the
 caller ID on the display of the person who answers to read Press 9 
 to open gate. During the call, they can hit 9 at any time and the 
 gate will open for them. Up until they answer, the caller ID reads 
 Gate Phone, but when they answer, it changes to that text. 
 
 The part about opening the gate is the magic piece you want to look 
 into. Read up on applicationmap in features.conf. It's pretty simple
 and very effective. Here is what mine looks like. I am going to 
 replace my actual command with insert command here. 
 
 gate = 9,self/callee,System,insert command here ; Custom 
 application to open the gate. 
 
 This says that this feature is active in the 'gate' context of my 
 dialplan. The dialing pattern it is looking for is a 9. 'self' tells
 it to activate on the channel that dialed it and callee says that 
 the person receiving the call is the only one that can activate it 
 (otherwise the person at the gate phone could hit 9 to open it). I 
 am running the System dialplan application and passing it the 
 insert command here value. Everything after the ';' is a comment 
 as normal. The insert command here is equivalent to what you would
 put inside the '()' if it were in the dialplan (i.e. 'System(insert
 command here)'). 
 
 Pretty straightforward to get it working once you know what to look 
 for. Let me know if you want to know how I manipulate the Caller ID 
 upon answering the call to give the instructions to the callee on 
 how to open the gate/door. 

I just realized I said one piece wrong in this. 'gate' is not the context, 
it is the dynamic feature designator. I can illustrate this better by 
posting my front gate context.

[front_gate]
exten = number gate dials goes here,1,Set(__DYNAMIC_FEATURES=gate)
  same = n,Goto(frontgate_queue,${EXTEN},1)

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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

 Deciding on the mailbox to use is problematic! The dialed-party may 
 be away for an extended period and wants voice mail handled by the 
 forwarded-to party.

And then you have the users who would work around this by sharing their 
voicemail passwords. Not quite as bad as sharing your computer log on 
credentials, but still, something I would like to avoid if possible.-- 
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Re: [asterisk-users] RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi Kevin.
 
 Thank you very much for the hint! It worked very well!
 
 Your example ' exten = 1234,1,System(echo This is a test  /
 var/log/asterisk/test.txt) ' executes when the SIP client (my 
 softphone Jitsi) sends  a SIP INVITE to asterisk.  So, the softphone
 tries to establish a session with target 1234.
 
 Now, lets suppose my softphone rings and I answer a call. During the
 call, the caller asks me to execute a command (ex: to open a door or
 gate). In this case, what have I to program in dial plan to Asterisk
 execute System() again? Is it possible to execute a dial plan even 
 during an ongoing call?
 
 Finally, lets suppose I want to use my softphone to execute a dial 
 plan, even without establishing a call (no session with target 
 1234). For example, If I decide to open a dor or gate using my 
 softphone, without existing an ongoing call, what have I to program 
 in dial plan to Asterisk executes System(). Is this idea possible?
 
 Any hint will be very hepful!

I love this question, simply because it allows me to talk about one of the 
neatest features I programmed into my system that barely anyone knows 
exists. Plus it lines up pretty much exactly with what you are trying to 
do.

We have our gate control system tied into our Asterisk phone system so it 
is possible to dial a code on the phone and open the entrance gate to let 
someone in after hours. Only problem is this happens so rarely that no one 
(myself included) ever remembered the code. Thus a search for a better 
way.

Now, when someone uses the gate phone to request entry, I change the 
caller ID on the display of the person who answers to read Press 9 to 
open gate. During the call, they can hit 9 at any time and the gate will 
open for them. Up until they answer, the caller ID reads Gate Phone, but 
when they answer, it changes to that text.

The part about opening the gate is the magic piece you want to look into. 
Read up on applicationmap in features.conf. It's pretty simple and very 
effective. Here is what mine looks like. I am going to replace my actual 
command with insert command here.

gate = 9,self/callee,System,insert command here ; Custom application to 
open the gate.

This says that this feature is active in the 'gate' context of my 
dialplan. The dialing pattern it is looking for is a 9. 'self' tells it to 
activate on the channel that dialed it and callee says that the person 
receiving the call is the only one that can activate it (otherwise the 
person at the gate phone could hit 9 to open it). I am running the System 
dialplan application and passing it the insert command here value. 
Everything after the ';' is a comment as normal. The insert command here 
is equivalent to what you would put inside the '()' if it were in the 
dialplan (i.e. 'System(insert command here)').

Pretty straightforward to get it working once you know what to look for. 
Let me know if you want to know how I manipulate the Caller ID upon 
answering the call to give the instructions to the callee on how to open 
the gate/door.-- 
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Re: [asterisk-users] RES: RES: RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi Kevin.
 
 Thank you again for help me!
 
 In my case,  in the final application for smartphones or in a 
 softphone for PCs, there will be a button on the GUI and the user 
 will have just to touch it, and the door or gate will open. I mean, 
 during an ongoing call, the callee will see a button in the 
 interface of its SIP application. For example, we can use the lib of
 Linphone and implement a GUI over it, having a new button to open 
 doors and gates. So, the callee will not have to remember about 
 codes, because there will be a button in someplace to be touched.
 
 When the button be touched, during an ongoing call, the software 
 (SIP client) will sends a request to Asterisk executes the gate = 
 9,self/callee,System,insert command here , for example. So, it 
 will works like the user pressing number 9.
 
 I will take a look at applicationmap in features.conf to understand 
 what exactly can be done.
 
 But, let me ask you:
 This idea seems to be good to run during ongoing calls. What about 
 moments when there is no ongoing call? That is, can Asterisk execute
 a dial plan (maybe by means of some kind of SIP request received 
 from the SIP client) even without establishing a call?

The way I would probably approach what you want to do is that the button 
action state would be dependent on if you are in a call or not. If you are 
in a call, it sends whatever DTMF digits you want to use for this feature. 
If you are not in a call, it could dial an extension whose purpose is to 
do the same thing. 

I have an outside number that when dialed checks that your caller id 
number is in an approved list and if it is, sends the gate open signal. 
This is the same gate open signal that the feature code uses (the call to 
System()), it is just reached by making a sip call. Nothing says a call 
has to connect two phones together. You can answer the call inside of 
Asterisk and do stuff based on what number you called or what digits the 
caller enters with their keypads. Lot's of opportunity to make the system 
do exactly what you want.-- 
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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

  Ia had a server overload today because someone did a call forward 
 to their own extension.  To do a call forward I write a key called CFWD 
 with the extensión number and number to dial .  The main script tests if 

 the key/value exists and dials the number stored in the database.  What 
 is an easy way to prevent dumb people from creating a loop?

Right after you have read the number to call forward to, compare it to the 
number you are call forwarding from. If it matches, play the user an error 
message and have them try again.

And no matter what you do, the dumb people will come up with more creative 
ways to tank your phone system. A large amount of my dialplan code is 
taking into account the stupid things they have done and handling it 
properly if they do it again. I swear, if you could harness their 
creativity for good you could solve the world's problems 10 times over.-- 
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Re: [asterisk-users] How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Hi everyone.
 
 I'm new with Asterisk and I have to create a dial plan that will 
 invoke a binary code. That is, asterisk will execute a program in 
 the same machine. How to do it?
 
 Let me explain what I have to do:
 
 In the project that I am currently working, there is smartphones, 
 SIP servers and doors/gates to be unlocked remotely. When the user 
 executes an application on his/her phone, it will presents a button 
 to unlock a remote gate or door.
 By  pressing such button, the application will send a SIP INVITE to 
 the SIP server (Asterisk). In this moment, a existing dial plan 
 should call an executable hosted in the current machine. In this 
 case I need to know how to program my extensions.conf to let 
 Asterisk invoke another software to me.
 The another software is the one responsible for unlocking a gate or 
door.
 
 So, how to codify my extensions.conf in order to make Asterisk 
 invoke another software?
 Is another better way (idea) to implement my project using Asterisk 
 and SIP? If so, comment, please!
 
 Any hint will be very helpful!

Look into the System() dialplan application. It will execute a command on 
the system for you. Be aware that it will execute it as the user your 
Asterisk instance is running as, so permissions can sometimes be a bit 
finicky to get correct. I do something similar to pop my gate open. It is 
using nc to make a connection to the device, but same general idea as what 
you are doing.-- 
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Re: [asterisk-users] RES: How to invoke a binary file from the dial plan?

[Kevin Larsen]

 Ok. Thanks for the hint.
 
 But, what exactly is a System() dialplan application? Is it a kind
 of command that i can call in dial plan? 
 
 I will look for System() related to dial plans.

From the Asterisk CLI type:
core show application System

It will print out the syntax for the command. One of the easier dialplan 
applications. 

exten = 1234,1,System(echo This is a test  
/var/log/asterisk/test.txt)

That line would use the Linux echo command to place the text This is a 
test into a file named test.txt located in the /var/log/asterisk 
directory.-- 
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Re: [asterisk-users] Forward loop protection...

[Kevin Larsen]

 The loop checking is a bit more challenging than that. If Bob 
 forwards to Fred and Fred forwards to Sue, all is well when Bob and 
 Fred head out for a beer. A little later, we’re in deep doo-do0 when
 Sue forwards to Bob. 

 Could this possibly mean that any person who has CF set should never
 be available as CF Destination. Simple db entry/check can have this 
done.

That just goes to show that the problem can get complex pretty quickly. 
Using the original example above, it might be that you want to allow the 
Bob to Fred to Sue forwards, but only stop it if the Sue to Bob link is 
established, thus creating the loop. I wonder if you could do some kind of 
recursive check where you follow each forward and if you ever come back 
around to a number you have already checked you know there is a loop.

To reuse the example above, on the creation of the Bob to Fred forward, 
the database is checked to see if Fred has any forwards. He doesn't, so is 
at the end of the forwarding chain. Now Fred forwards to Sue. Again, she 
is at the end of the chain, so it is allowed. When Sue goes to forward to 
Bob, the check shows that Bob has a forward. Not a problem, but we create 
a temporary list that has Sue's number in it. Then we check the next stage 
of forwarding. Bob forwards to Fred. Fred's is checked against our 
temporary list and doesn't match, so we are still good. Bob's number is 
now added to the temporary list and we check the forward Fred has in 
place. Fred forward's to Sue. We check Sue's number against the temporary 
list and it does exist. Thus we have a loop detected and the forward can 
now be denied.

I am guessing with the recursion involved you might want to do the check 
outside of Asterisk and pass the result back in. I will also state that I 
have not had to do this deep checking in the past, so these are just some 
initial thoughts on how I would start approaching the problem. Of course, 
this also assumes that Bob, Fred, and Sue are all on the same phone 
system. If you don't have a shared database to look at, the problem just 
got harder indeed.
-- 
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Re: [asterisk-users] Signaling incoming call

[Kevin Larsen]

 Hi Kevin!
 
 Thanks! It works!
 I can set the name of the line with CALLERID(name) and see the caller 
number,
 too.
 And, it the number is in the address book, I see the name, too.
 
 Perfect!

Glad it worked for you. I usually leave the number untouched, but will 
manipulate the name to suite what I want. I have mulitple call queues, so 
for instance, for my helpdesk lines, I will do something like transform 
Name to HD:Name so that the person being called knows that the caller 
dialed the help desk number rather than their direct number. On people who 
work multiple queues, it is very handy so they can see at a glance what 
queue the caller is reaching.-- 
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Re: [asterisk-users] Signaling incoming call

[Kevin Larsen]

 Hi Steve!
 
 Thank you very much!
 It seems to run!
 
 I wrote that:
 
 exten = _0049351333,n,Set(__ALERT_INFO=Bellcore-r3)
 exten = _0049351333,n,SIPAddHeader(Alert-Info:
http://www.notused.com
 \;info=alert-external\;x-line-id=0)
 
 and the phone rings with another melody.
 Very curious is, that if I don't write BOTH lines, it does not run...
 
 And, unfortunately, I just have two melody: the normal and this one, 
but it
 is better than nothing!
 Now, if it will be possible to add a text on the display, it will be 
perfect,
 but I didn't found any option for that...

Look into Set(CALLERID(name)) and Set(CALLERID(num)) to manipulate the 
caller id name and number that show up on the phone.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

  What kind of phone are we talking about, both yours that works and 
your 
  wife's that does not?
 
 Right!
 
  Can you ping the unreachable phone and does it respond to a ping?
 
 I can ping both phones from the VM
 
  Many phones will have a network test function built in to them to help 
you 
  determine if the phone is properly connected to the network.
 
 Unfortunately not that...
 I tried with Twinkle from my PC, using the same account of my wife
 (configured IDENTICALLY to my account, just another username). It don't
 work...
 I presume, I configured something wrong in Asterisk...
 
  Do you see anything in the asterisk logs or the logs of the phone 
itself 
  (providing the phone puts logs somewhere) that indicate a failure to 
  register or to resolve the ip address of the asterisk server?
 
 Unfortunately not... Just UNREACHABLE...

Can you post the Manufacturer and Model of your phones (both of them if 
they are different)? That will help us look up what diagnostics/log files 
there might be on the phones.

Does the Twinkle software on the PC show any error messages?

If you watch the CLI in asterisk, does anything go by in there regarding a 
failed registration? If I get one of my phones programmed with an 
incorrect username/secret, it will try to register with the server, but 
can't. Those failed registrations do show up in the CLI.

Double check that you are not mistyping the credentials somewhere. If you 
do post the relevant parts of your config in here, you might want to 
obscure the secret.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 I have a problem and I hope someone can help me...
 I configured an Asterisk on a VM to serve more accounts and act as a 
proxy to
 other SIP-providers.
 
 The first account running on my phone works without any problem.
 A second account, running on the phone of my wife, is always 
UNREACHABLE.
 I can just see in the log:
 
 [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer
 '004935111' is now UNREACHABLE!  Last qualify: 0
 
 In the CLI I can see:
 
 Name/username  HostDyn Nat ACL Port Status  
 004935111/0049351  192.168.200.11   D  5060 UNREACHABLE 
 004935122/0049351  192.168.200.10   D  5060 OK (17 
ms) 
 004935133  (Unspecified)D  5060 UNKNOWN  
 
 1234   (Unspecified)D  5060 UNKNOWN  
 
 messagenet/1234567890  212.97.59.765061 Unmonitored 
 pbxanika/004935172.16.34.132   5060 Unmonitored 
 pbxfax/0049351333  172.16.34.132   5060 Unmonitored 
 pbxluca/0049351222 172.16.34.132   5060 Unmonitored 
 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 
offline]
 
 Asterisk connects to another Test-VM with AsteriskNOW and to the italian
 provider Messagenet.
 
 Can someone suggest me, what can I do?
 I can send the configuration file, if they are needed.
 

What kind of phone are we talking about, both yours that works and your 
wife's that does not?

Can you ping the unreachable phone and does it respond to a ping?

Many phones will have a network test function built in to them to help you 
determine if the phone is properly connected to the network.

Do you see anything in the asterisk logs or the logs of the phone itself 
(providing the phone puts logs somewhere) that indicate a failure to 
register or to resolve the ip address of the asterisk server?-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 Darryl Moore dar...@moores.ca schrieb:
 
  I'd start by turning on sip debugging in asterisk
   sip set debug ip [your_phone_ip]
 
 Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.
 16.34.133' Method: OPTIONS
 Reliably Transmitting (no NAT) to 192.168.200.11:5060:
 OPTIONS sip:0049351222@192.168.200.11:5060 SIP/2.0
 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport
 Max-Forwards: 70
 From: asterisk sip:asterisk@172.16.34.133;tag=as1215345d
 To: sip:0049351222@192.168.200.11:5060
 Contact: sip:asterisk@172.16.34.133
 Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6@172.16.34.133
 CSeq: 102 OPTIONS
 User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
 Date: Thu, 28 May 2015 20:39:02 GMT
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Length: 0
 
 repeated in loop...
 Help that?
 
 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 
 the IP of the Asterisk server.
 

The phone you gave your wife is really old. Are you sure it supports SIP 
OPTIONS? Can you make a call in or out to it? If you can, it is more 
likely that it just doesn't support that and you can't use a qualify 
statement.-- 
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Re: [asterisk-users] Peer is UNREACHABLE

[Kevin Larsen]

 No, I'm not sure.
 And no, I can't make any call, right now... At least, not connected to 
my
 Asterisk...
 If I connect it to the other VM with AsteriskNOW I can call my Twinkle, 
but
 NOT my phone connected on my Asterisk, using the proxy.
 I can see that in the log:
 
 [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username
 mismatch, have 1234, digest has luca
 [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite:
 Failed to authenticate device Test1 
sip:1234@172.16.34.132;tag=as6dd12e05
 

I know from your previous email that you are new to Asterisk. Have you 
created a dialplan that would allow you to call from one extension to 
another without going through your phone company? That is to say, call 
from your phone through Asterisk to your wife's phone?

You have two parts that you need to have in place for the basics to work. 
You need your sip.conf in order to tell asterisk what devices and phone 
trunks you have and you need extensions.conf to tell Asterisk how to route 
calls. Since you are new to this, you can start by getting the two phones 
to both register (sounds like one of them is and one probably is not). 
Then you get to where you can dial from one phone to the other and vice 
versa. From there you can add in the telephone company lines and the 
ability to dial in and out to the world.

I am still curious why you have both an Asterisk setup and an AsteriskNow 
setup? Is that just to play around with? At the end of the day you should 
just need one or the other.-- 
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Re: [asterisk-users] Asterisk as Proxy and more device for a number

[Kevin Larsen]

 I'm very new in Asterisk and VoIP, and of course I have a problem... :)
 
 Well, my problem is, that Deutsche Telekom wants me to change my ISDN 
 to VoIP... :(
 I must do that, since I have no alternative.
 
 Well, I have now two VoIP-phones (Thomson ST2022 and KE1020A). I can 
 configure my two numbers by Deutsche Telekom and I got now an extra 
 number from Messagenet.it.
 
 Now the problems:
 1) It seems that I can't configure my ST2022 to have two profiles and 
 both are running on different servers
 2) I want that when a number will be called, both phones rings
 
 I think, I need an Asterisk-Server between my phones and the 
 VoIP-Provider, isn't it?
 
 Well, now the questions: am I right? Should I install an Asterisk on 
 my PC to do that?
 And of course: how can I do that? How can I set up Asterisk to serve 
 as proxy for these three numbers and send the calls to a number to 
 both phones?
 
 Unfortunately, I didn't found any HowTo for my problems...
 

If you want to go the Asterisk from scratch route, you would do well to 
pick up a book on the subject. Since you seem comfortable with English, 
Asterisk: The Definitive Guide is a good place to start. This will teach 
you how to build an Asterisk system from the ground up. Depending on what 
you want to do, this may also be overkill.

There are Asterisk distributions that already come with a GUI front end 
that could make this all a lot easier to set up. AsteriskNow (includes 
Asterisk and FreePBX Gui) is a good choice as would be Elastix (Asterisk + 
FreePBX GUI + the Elastix GUI). These are often much easier to set up for 
the Asterisk newbie. Either of those should be able to easily handle what 
you want to do.

Even if you do go the route of a pre-made distribution with a GUI, the 
Asterisk book is still useful to have. It really gives great insight into 
the software and will help if you ever have to troubleshoot from the 
command line.-- 
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Re: [asterisk-users] Asterisk as Proxy and more device for a number

[Kevin Larsen]

 Maybe I got it...
 I installed an asterisk on a VM with Ubuntu 10.04 and I got it 
connecting to
 another Test-VM with AsteriskNOW and with an italian VoIP-provider.
 The very difficult was to understand, that my phone just can manage ONE
 profile at time, so I had to configure Asterisk to receive all calls 
from the
 different providers an send them ONE profile (on my phone).
 
 Next step is to configure Asterisk for the other phone (for my wife) and
 having all calls of her number forwarded to my phone and her phone.
 
 Next step again is to manage outgoing calls going to the right provider.
 
 Then it would be nice if I can forward calls from a phone to the other.
 
 Last but not least, I need to use HylaFAX on an account on Asterisk.
 I had many problems with T38Modem, so I'll try with IAXModem, maybe I'll 
got
 it...

Glad you have it working. You should only need one Asterisk server to do 
what you want unless you just want to have one with the GUI and one for 
testing purposes. I would recommend starting with something newer than 
Ubuntu 10.04 as it is pretty much at its end of life. 14.04 would be a 
better choice at this point.

Regardless of how you end up directing your incoming calls, that KE1020A 
phone is pretty old and it might be worthwhile to see about upgrading it 
to something newer. The Thomson ST2022 you have does seem to have the 
capability to have two lines on it. Haven't used one before, so hard to 
say how good it handles that. Whatever you do, though, having two 
identical phones will be helpful to you (and your wife) as you won't have 
to try to remember how each phone works and troubleshooting problems is 
easier if you can look at a phone that is working of the same model.

There are a couple of ways you can approach directing your calls to the 
right outgoing provider. One would be to have two separate lines on your 
phone and just pick which one you want to use that will direct all calls 
to the right provider. If your calls follow a pattern (i.e. calls to this 
country go to this provider and calls to that country to to the other 
provider), you can have Asterisk recognize the pattern and automatically 
direct the calls for you. This is nice as others won't have to remember 
which line to use.

Asterisk has built in forwarding capabilities by dialing the right feature 
code during a call to initiate a forward to another extension. Many phones 
also have this feature built in. I use Polycom phones and can transfer 
calls just by hitting the transfer button and dialing who I want to 
transfer to.

I have used the HylaFax/IAXModem solution with a client and it worked 
fairly well. I will warn you that faxes over VoIP connections are 
inherently worse than over a regular phone line. They can be made to be 
almost as good or they can just be horrible, but either way, faxing is no 
fun, especially considering that the problems can be caused before the fax 
ever reaches your system. Hopefully your provider supports T.38 properly, 
in which case faxing will be much nicer.-- 
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Re: [asterisk-users] Phone provisioning template Snoms

[Kevin Larsen]

  I am looking for a phone provisioning template for Snom phones, 
 Yealinks and Polycoms. I am always doing deployments of many phones 
 and usually configure each phone one by one for each installation. 
 Any help will be highly appreciated
 
 There’s some excellent documentation about provisioning on the Snom 
Wiki:
 
 http://wiki.snom.com/Category:Auto_Provisioning:Configuration_Files
 
 You can set the phones (via DHCP options) a firmware url on a web 
 server under your control, grab their MAC addresses, then deliver 
 them custom config settings as required.
 
 Easiest way to start is to copy the config file (via the web 
 interface) from a phone with factory default settings, then just 
 change the settings you need to change, and write something in your 
 scripting language of choice (PHP, Perl, Python, etc.) to just send 
 those settings to the phone dependent on MAC address. Don’t send 
 *every* available config setting to the phone - only the changes 
 from default you need to make.
 
 I suspect the same can be done with Yealink and Polycom phones - 
 I’ve not used those so can’t really comment. I have a similar system
 which seems to work for Sipura/Linksys/Cisco phones, though most of 
 my new deployments are exclusively Snom.
 

I use the Polycom phones and there are any number of ways you can automate 
deployments of them. The templates you want to start with can be found on 
the Polycom website here: 
http://support.polycom.com/PolycomService/support/us/support/voice/index.html

When you download the firmware (UC software release), you get the 
templates you want included in the download. You can use FTP, TFTP, and 
HTTP that I know of to provision the phones. I use HTTP and have some 
custom php scripts that I wrote that create my own templates on the fly 
for a phone based on its mac address. You can use a combination of static 
templates for things that are system wide and dynamic templates for the 
things that are specific to each phone. You can also just create a static 
template for each phone. It really depends on how in depth you want to go. 
It does make provisioning much easier though.
-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

[Kevin Larsen]

 I am using 11.17.0 -  and MulticastRTP. Doesnt seem to work with 
 polycom phones as other devices receive my multicast just fine.
 
 Is there something special to do to get multicast working with polycom 
phones?
 (other than enable multicast on the actual phone).

Didn't see if anyone had answered you or not on this, but Polycom uses 
their own form of MulticastRTP. It doesn't work with Asterisk's multicast 
setup. There is a company that makes a loud ringer/pager unit that can 
also be used to take in a sip call and multicast out to the Polycom 
phones. I haven't tried it myself as I just use the loud ringer 
capability, but it does appear that it would be a workable solution. I 
hesitate to promote the name here since this is non-commercial discussion, 
but let me know if you want to hear the actual product.-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

[Kevin Larsen]

  I hesitate to promote the name here since this is non-commercial 
  discussion...
 
  but Polycom...
 
  Polycom phones...
 
 If mentioning Polycom is OK, I think mentioning a possible commercial 
 solution is OK.

In that case, the product in question is the Algo 8180 SIP Audio Alerter. 
I will state that I have not used this particular functionality, but it is 
mentioned in the users guide, so in theory you could use it as a bridge 
between Asterisk and mulitcast on the Polycom phones. YMMV.-- 
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Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM:
 I'm looking at enabling autopause on one of my queues where my queue
 members are bad about leaving their desks without pausing.

 The problem I see is that when the queue pauses an Member it doesn't
 jump into the dialplan to do so which means my handy device state 
 and asterisk database driven Light for the Member showing their 
 paused status won't update.

 My idea for solving this problem is to check the status of my Member
 in the queue before I send the calls into it and toggle on the 
 Members Paused light at that point in time if they are paused.

 Sadly I don't see a way to determine if my Staff are paused or not 
 from the dialplan, There doesn't appear to be a function to retrieve
 the status of the members in the queue.

 Does the list have any suggestions?

First, let me say I feel dirty for even posting this. It is probably far 
from ideal, but it does get the job done. I had the same issue. Also, I am 
using Asterisk 11. I just looked and it doesn't appear that the 
QUEUE_MEMBER function supports the paused option in 1.8. To be honest, I 
am not sure if there is a good replacement for what I have done below in 
the 1.8 series.

[sub_autopause_status]
exten = s,1,NoOp(Checking for autopaused members for ${arg1} queue)
  same = n,Set(MEMBERS=${QUEUE_MEMBER_LIST(${arg1})})
  same = n,Set(i=1)
  same = n,Set(max=${FIELDQTY(MEMBERS,,)})

  same = n,While($[${i} = ${max}])
  same = n,Set(MEMBER=${CUT(MEMBERS,\,,${i})})
  same = n,Set(STATUS=${QUEUE_MEMBER(${arg1},paused,${MEMBER})})
  same = n,Set(MEMBER_EXT=${CUT(MEMBER,\/,2)})
  same = n,ExecIf($[${STATUS} = 0]?System(echo IN  
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
  same = n,ExecIf($[${STATUS} = 1]?System(echo PAU  
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
  same = n,NoOp(${MEMBER}: ${STATUS})
  same = n,Set(i=$[${i} + 1])
  same = n,EndWhile()

  same = n,Return()


So, as an explanation, I have multiple queues and agents who autopause. I 
show their status on their phones, hence the System(echo...) commands to 
the /var/spool/asterisk/status directory. Those files are used to generate 
a simple web page that is shown on their phones that lets them see their 
status. You should be able to adapt that to what you do.

Basically, you pass the queue name into the subroutine as arg1. The 
subroutine gets a list of every person logged into that queue and then 
loops through checking the status of each person using the QUEUE_MEMBER 
function.

It isn't elegant and if you have a lot of queues/queue members to check, 
it will constitute a lot of looping, but it does work. Like you, I would 
like to have a way to check the pause status of a member easier. If the 
queue application could call a subroutine with it autopaused someone, that 
would actually make an elegant solution, but for now, this was the way I 
could see to do it.

You could maybe call a script that would parse the queue_log file looking 
for an agents status and pass that back into the dialplan.-- 
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Re: [asterisk-users] [OT] switches

[Kevin Larsen]

 so how does a client pc find the server if there's no NAT?  by IP 
 address?? That makes no sense, to me, if the switch isn't assigning 
 addresses.

Switches have a MAC table that keeps track of which MAC addresses are on 
which ports. That's how they decide where to route packets.

http://en.wikipedia.org/wiki/CAM_Table
http://en.wikipedia.org/wiki/OSI_model-- 
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Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value uri.

[Kevin Harwell]

On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com wrote:

  Hello.

 Asterisk 13.2.
 I transfer configs from chan_sip to res_pjsip.
 In chan_sip i have match_auth_username=yes and have nothing in pjsip.

 I have a lot of endpoints and registrations on same SIP server. And it's
 problem in pjsip now. Is not it?

 I requesting to add new value for endpoint option identify_by. The value
 'uri'.
 Simple config (cutted):

 [siptrunk]
 type=registration
 transport=udp-transport
 outbound_auth=siptrunk
 server_uri=sip:sip.example.comclient_uri=sip:1234567...@sip.example.com
 retry_interval=60
 contact_user=siptrunk-in

 [siptrunk-in]
 type=endpoint
 transport=udp-transport
 context=from-trunk
 disallow=all
 allow=ulaw
 outbound_auth=siptrunk
 aors=siptrunk
 identify_by=uri


 Registration section has option contact_user. Incoming call from this
 registration will be INVITE sip:siptrunk-in@
 I offer to change res_pjsip_endpoint_identifier_user to realize endpoint
 identification by sip uri.

 I think it will be usefull.

 P.S. i hope issues will be rejected:
 https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069


 Dmitriy Serov

 -


I believe what you are looking for is already available. See the identify
type (type=identify) section that is in the pjsip.conf file and the
identify option for endpoints. These allow you to identify and endpoint
by IP address.

For more information see the pjsip.conf.sample file.  Also take a look at
configuring Asterisk for res_pjsip [1] specifically the part about
configuring endpoint identification by IP address [2]. If you run into
problems more information can also be found in the res_pjsip
troubleshooting guide [3], specifically the section on identify by IP
address

[1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
[2]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip
[3]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide

Hope that helps,

-- 

Kevin Harwell
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org
-- 
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