from:"Saint Michael"

Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 17

2021-01-24 Thread Saint Michael

Re: Get a SHAKEN Identity Token (Alexander Perkins)

Saint Michael 
1:06 PM (0 minutes ago)
to Asterisk
Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail service.


On Sun, Jan 24, 2021 at 1:00 PM 
wrote:

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>1. Re: Get a SHAKEN Identity Token (Alexander Perkins)
>
>
> --
>
> Message: 1
> Date: Sat, 23 Jan 2021 20:30:42 -0500
> From: Alexander Perkins 
> To: Markus 
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> callktp0j5hfj1ou+rhrxaevab_wvxqzcoon9s-kfpvzwt6m...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Markus.  Thanks a bunch!  I will try that out!
>
> On Fri, Jan 22, 2021 at 8:06 AM Markus  wrote:
>
> > Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
> > > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> > > of several reasons) and we are now implementing SHAKEN via our
> > > provider.  We place a SIP call to our provider and they return a 302
> > > (information below).  I am trying to get the X-Identity information
> > > below, but I do not seem to be able to do so.  Can somebody help me
> with
> > > this?  Any suggestions on how to get it?
> >
> > I use SIP_HEADER to extract information from inbound SIP packets and
> > SIPAddHeader to copy that info to the outgoing call leg. Maybe this
> > helps you?
> >
> > Example:
> >
> > exten => _+X.,1,NoOp(${CALLERID(num)})
> > exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
> > exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
> > exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
> > exten => _+X.,n,Goto(dial)
> > exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: id)
> > exten => _+X.,n,Goto(dial)
> >
> >
> >
> -- next part --
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Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

2021-01-24 Thread Saint Michael

Please look at this
https://issues.asterisk.org/jira/browse/ASTERISK-28924
I have a solution that works for any version of Asterisk, if interested
contact me at venefax at the Google mail service.

On Sun, Jan 24, 2021 at 1:00 PM 
wrote:

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> Today's Topics:
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>1. Re: Get a SHAKEN Identity Token (Alexander Perkins)
>
>
> --
>
> Message: 1
> Date: Sat, 23 Jan 2021 20:30:42 -0500
> From: Alexander Perkins 
> To: Markus 
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> callktp0j5hfj1ou+rhrxaevab_wvxqzcoon9s-kfpvzwt6m...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Markus.  Thanks a bunch!  I will try that out!
>
> On Fri, Jan 22, 2021 at 8:06 AM Markus  wrote:
>
> > Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
> > > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> > > of several reasons) and we are now implementing SHAKEN via our
> > > provider.  We place a SIP call to our provider and they return a 302
> > > (information below).  I am trying to get the X-Identity information
> > > below, but I do not seem to be able to do so.  Can somebody help me
> with
> > > this?  Any suggestions on how to get it?
> >
> > I use SIP_HEADER to extract information from inbound SIP packets and
> > SIPAddHeader to copy that info to the outgoing call leg. Maybe this
> > helps you?
> >
> > Example:
> >
> > exten => _+X.,1,NoOp(${CALLERID(num)})
> > exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
> > exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
> > exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
> > exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
> > exten => _+X.,n,Goto(dial)
> > exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
> > )
> > exten => _+X.,n,SIPAddHeader(Privacy: id)
> > exten => _+X.,n,Goto(dial)
> >
> >
> >
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Re: [asterisk-users] asterisk-users Digest, Vol 197, Issue 7

2021-01-08 Thread Saint Michael

Stir Shaken
Asterisk cannot do that, but my company can give you Stir Shaken for
Asterisk, via ODBC, any version.
Please contact me via email venefax at the google mail system
Philip Orleans

On Fri, Jan 8, 2021 at 1:00 PM 
wrote:

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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Get a SHAKEN Identity Token (Alexander Perkins)
>2. Re: Get a SHAKEN Identity Token (Joshua C. Colp)
>
>
> --
>
> Message: 1
> Date: Thu, 7 Jan 2021 17:49:27 -0500
> From: Alexander Perkins 
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
>  hxaeuyw+lx0guee9smj-0su1hqge8s...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because of
> several reasons) and we are now implementing SHAKEN via our provider.  We
> place a SIP call to our provider and they return a 302 (information
> below).  I am trying to get the X-Identity information below, but I do not
> seem to be able to do so.  Can somebody help me with this?  Any suggestions
> on how to get it?
>
> Thank you, All.  Very much appreciated!
>
> <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
> SIP/2.0 302 STIR/SHAKEN
> Via: SIP/2.0/UDP
>
> XXX.XXX.XXX.XXX:5066;received=XXX.XXX.XXX.XXX;branch=z9hG4bK37b49c97;rport=5066
> From: "12125551212" ;tag=as0026c4e3
> To:  >;tag=bcaa-20103108495689bb4065d39c43badb69
> Call-ID: 6755a4484427f12b0e56d6903fe50...@xxx.xxx.xxx.xxx:5066
> CSeq: 102 INVITE
> X-Identity:
>
> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0aWZpY2F0ZXMuY2xlYXJpcC5jb20vZmRmYjMzMjgtYjc1NC00YTBkLThiMzQtZGUzMGIwOGFkYWMyLzQ3NmMyODliYTgxY2QxNWU3MjBmNzkxOWM5NGU5MzU2LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjU2NTI3NjIwMSJdfSwiaWF0IjoxNjEwMDU2MDE3LCJvcmlnIjp7InRuIjoiMTI1NjkwNjQ5NTUifSwib3JpZ2lkIjoiMGFlODFjZWQtYzhlZS00ZWFiLTliNjAtMDY3OWM0Y2Q1MjUwIn0.lr3uj0fmlHbSori-msdbvKu5SQrVnLA-ZMswCY_dLk79jrpr1yFhWmL4GiAr16VtMKVSamQ-0bi3Pptoi7TUfw;info=<
>
> https://certificates.clearip.com/fdfb3328-b754-4a0d-8b34-de30b08adac2/476c289ba81cd15e720f7919c94e9356.crt
> >;alg=ES256;ppt=shaken
> Server: TILTX Technology Innovation Lab SHAKEN
> Content-Length: 0
> -- next part --
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> >
>
> --
>
> Message: 2
> Date: Thu, 7 Jan 2021 19:00:13 -0400
> From: "Joshua C. Colp" 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Get a SHAKEN Identity Token
> Message-ID:
> <
> cam0a2z2fyurcpi7_bh3gzjhkvx2jlqqcbxnzcrhhulbbbhj...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> On Thu, Jan 7, 2021 at 6:50 PM Alexander Perkins <
> alexanderhenryperk...@gmail.com> wrote:
>
> > Hi All.  We have old Asterisk servers, 1,89, (we cannot upgrade because
> of
> > several reasons) and we are now implementing SHAKEN via our provider.  We
> > place a SIP call to our provider and they return a 302 (information
> > below).  I am trying to get the X-Identity information below, but I do
> not
> > seem to be able to do so.  Can somebody help me with this?  Any
> suggestions
> > on how to get it?
> >
> > Thank you, All.  Very much appreciated!
> >
> > <--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
> > SIP/2.0 302 STIR/SHAKEN
> > Via: SIP/2.0/UDP
> >
> XXX.XXX.XXX.XXX:5066;received=XXX.XXX.XXX.XXX;branch=z9hG4bK37b49c97;rport=5066
> > From: "12125551212"  :5066>;tag=as0026c4e3
> > To:  > >;tag=bcaa-20103108495689bb4065d39c43badb69
> > Call-ID: 6755a4484427f12b0e56d6903fe50...@xxx.xxx.xxx.xxx:5066
> > CSeq: 102 INVITE
> > X-Identity:
> >
> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0aWZpY2F0ZXMuY2xlYXJpcC5jb20vZmRmYjMzMjgtYjc1NC00YTBkLThiMzQtZGUzMGIwOGFkYWMyLzQ3NmMyODliYTgxY2QxNWU3MjBmNzkxOWM5NGU5MzU2LmNydCJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjU2NTI3NjIwMSJdfSwiaWF0IjoxNjEwMDU2MDE3LCJvcmlnIjp7InRuIjoiMTI1NjkwNjQ5NTUifSwib3JpZ2lkIjoiMGFlODFjZWQtYzhlZS00ZWFiLTliNjAtMDY3OWM0Y2Q1MjUwIn0.lr3uj0fmlHbSori-msdbvKu5SQrVnLA-ZMswCY_dLk79jrpr1yFhWmL4GiAr16VtMKVSamQ-0bi3Pptoi7TUfw;info=<
> >
> https://certificates.clearip.com/fdfb3328-b754-4a0d-8b34-de30b08adac2/476c289ba81cd15e720f7919c94e9356.crt
> > >;alg=ES256;ppt=shaken
> >

Re: [asterisk-users] asterisk-users Digest, Vol 193, Issue 15

2020-09-26 Thread Saint Michael

memory vs disk cache

> This is an issue that has plagued Asterisk since day one. Basically there
>> is no solution available because there is no way to set aside memory to be
>> kept from a growing disk cache. I did some research and this looks like a
>> bad design from the Kernel people. Meanwhile all you can do us every 60
>> seconds:
>
> echo 3 | sudo tee /proc/sys/vm/drop_caches
> Asterisk should be able to reserve memory and force it to stay locked in
> memory, exactly like Mariadb does with
> memlock=1
> Would the Asterisk developers consider something like this?
>
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[asterisk-users] 1. memory issues (hw)

2020-09-26 Thread Saint Michael

>
> This is an issue that has plagued Asterisk since day one. Basically there
> is no solution available because there is no way to set aside memory to be
> kept from a growing disk cache. I did some research and this looks like a
> bad design from the Kernel people. Meanwhile all you can do us every 60
> seconds:

echo 3 | sudo tee /proc/sys/vm/drop_caches
Asterisk should be able to reserve memory and force it to stay locked in
memory, exactly like Mariadb does with
memlock=1
Would the Asterisk developers consider something like this?



>
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[asterisk-users] Stir Shaken

2020-07-14 Thread Saint Michael

I need to point out the this is factually misleading and materially false:
"I think this, being the basis of your whole argument, is the fallacy.
S/S is forcing people to take responsibility, for sure, but carriers
won't just let their customers leave because they don't want to sign
calls.  It will force them to make sure they know who their customers
are, and make it impossible for those customers to escape consequences if
they misbehave."

There is Law of The Land that is about to take effect. Use google and
search "stir shaken" Whoever thinks I am exaggerated is dreaming. Also: it
is true that my service is the only one for asterisk --worldwide. The model
proposed by Transexus (302 redirect with a new header) can't be used by
Asterisk.
But don't take my word for it:
https://issues.asterisk.org/jira/browse/ASTERISK-28924




>
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[asterisk-users] Stir Shaken

2020-07-13 Thread Saint Michael

>
> There is a big confusion here about Stir Shaken. It is NOT a provider
> issue. Un fact, all providers are whasing their hands and modifying their
> swihtches to pass-through the Signature. They cannot sign the call because
> then the become the responsible party for the call before the FCC, and
> liable for any illegal call. Every owner of a PBX that sends calls to the
> network, except if you use a trunk for the likes of Vonage, needs to sign
> their calls. So if you send calls with any kind of dialer and use DIDs,
> real or "borrowed", you need to get the signature service urgently or your
> business will stop terminating calls. You cannot self-sign, you cannot get
> around it, the calls will either go to straight to voicemail or fail. Even
> worse, the carries wil play a fake voicemail and charge you a fee,
> something that some already a are doing when they detect robocallig.

Don't even think about Transnexus, because they use 302 Redirect with a
header, and no version of Asterisk supports it.  I am the only game in the
world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is
literally true. If you need to sign your calls to get through, with
Asterisk, you need to connect to my service. I am an approved Service
Provider from the FCC. If you keep thinking this is not happening, it is,
and your business will disappear overnight.
The issue is that Vicidial, for example, does not provide res_odbc and
func_odbc, so you need to solve that first with Vicidial. Then you can
apply the code I provided earlier and your calls with have a legal, binding
signature. The carriers verify each signature and discard the ones that
fail the cryptography test.
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[asterisk-users] Stir Shaken is upon us

2020-07-12 Thread Saint Michael

WORLDWIDE EMERGENCY
The code below needs to be executed before any SIP or PJSIP call destined
to the US network, or soon no call will terminate. This is called
Stir-Shaken, a new law from the FCC.
If this is not working the whole Asterisk industry will crash, vanish, be
gone. I am assuming that the caller ID and the Destination Number are in
the variables "${CALLERID(num):-10}" "${EXTEN:-11}"

;Dialplan section to execute before any Dial
[strshk]
exten =>
_X.,1,Set(ARRAY(Token)=${MYSQL_STRSHK(${CALLERID(num):-10},${EXTEN:-11})})
;same=n,Verbose(0,Token ${Token})
;same=n,SIPAddHeader(Identity:${Token}) ;OLD SIP CHANNEL
same=n,Set(PJSIP_HEADER(add,Identity)=${Token}) ; NEW PJSIP CHANNEL
 same=n,Return()

/etc/odbcinst.ini or /etc/unixODBC/odbcinst.ini
[ODBC]
Trace=No
Trace File=/tmp/sql.log
Pooling=yes

[maria]
Description=ODBC for MySQL
Driver=/usr/lib64/libmaodbc.so
FileUsage=1
Threading=0

/etc/odbc.ini or /etc/unixODBC/odbc.ini
[strshk]
Description = MySQL ODBC Driver Testing
Driver = maria
Server = 208.73.232.47
#free testing service
User = anonymous
Password =
Database = strshk
Option = 3

res_odbc.conf
[strshk]
enabled=yes
dsn=strshk
sanitysql => select 1
isolation => read_uncommitted
username=anonymous
password=
pre-connect => yes
forcecommit => yes
connect_timeout => 10
negative_connection_cache => 300
max_connections=100
database=strshk

func_odbc.conf
[STRSHK]
escapecommas=yes
prefix=MYSQL
dsn=strshk
readsql=call strshk.stir_shaken_signature('${ARG1}','${ARG2}')
escapecommas=yes

Of course, you need to compile the modules res_odbc and func_odbc, which I
have done for Vicidial using Asterisk 13. But any Asterisk 11 and up can
use unixODBC.
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[asterisk-users] Extracting a SIP Header from a 302 Response

2020-05-30 Thread Saint Michael

I got the response below from a provider. How do I extract the Identity
header and apply it to the next INVITE? Is it possible at all with PJSIP?
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 172.16.7.254:52169
;rport=52169;received=XX.205.172.89;branch=z9hG4bK-524287-1---129f4244aaba9f04
Call-ID: 102650Mzg4NmFiNTQzOGY5NDJmNjM3OTYzNmE5MzNlZDIwZmI
From: "Peter Perez" ;tag=81a25c36
To: ;tag=9e198dc4-7ce8-433d-ae23-05b9bc14d55a
Identity:
eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cDovL2NlcnQtYXV0aC5wb2Muc3lzLmNbWNhc3QubmV0L2V4YW1wbGUuY2VydCJ9eyJhdHRlc3QiOiJBIiwiZGVzdC6eyJ0biI6IisxMjE1NTU1MTIxMyJ9LCJpYXQiOiIxNDcxMzc1NDE4Iiwib3JpZyI6eyJ0biI64oCdKzEyMTU1NTUxMjEyIn0sIm9yaWdpZCI6IjEyM2U0NTY3LWU4OWItMTJkMy1hNDU2LTQyNjY1NTQ0MDAwMCJ9._28kAwRWnheXyA6nY4MvmK5JKHZH9hSYkWI4g75mnq9Tj2lW4WPm0PlvudoGaj7wM5XujZUTb_3MA4modoDtCA
;info=;alg=ES256  CSeq: 1 INVITE
Server: Asterisk PBX 16.10.0
Contact: 
Reason: Q.850;cause=0
Content-Length:  0
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[asterisk-users] Stir-Shaken clarified

2020-05-29 Thread Saint Michael

https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN
The Wiki above is misleading in what Stir-Shaken means and how it works.
End users cannot get a certificate, they cannot self-certify their calls.
Somebody completely misunderstood the model. I am afraid the moment will
come and thousands of Asterisk operators will be unable to terminate calls.
To start with, the model is a hierarchical one: there is an FCC
designated central authority, which appoints (so far two) Certification
Authorities, allowed to issue Certificates for Service Providers ONLY,
which themselves are ALSO pre-approved by then GA (Governance Authority),
and they need to have an OCN, they need to be a CLEC, have their own block
of numbers. So the idea that an Asterisk operator can have its own
certificate and somehow calculate the signature, is ridiculous. Once the
call arrives a the last mile, let's say VZ or ATT, the carrier will open
the signature added to each call and verify it with the Certification
Authority that issued the certificate. They will check if the caller-ID and
destination number match the actual call. Each signature is valid only for
60 seconds and each call has a different signature, even for the same
caller-ID and destination number, so it cannot be stored.
As you can see, this is a new world and we need to prepare for its arrival,
or our calls will simply fail and we shall be out of business. My company
is an approved Service Provider and we are waiting for the certificate,
which is in itself complicated paperwork.
Our model to solve this riddle for Asterisk is simple: Add a
res_odbc.so-connection pointed to our MySQL database. Create a func_odbc
function that executes our stored procedure. For each call, you send us the
pair Caller-ID and Destination number, and we send you back the signature.
In the next line in the dialplan, you add a SIP-header called Identity, and
our signature becomes the content.
Identity:
eyJhbGciOiJFUzI1NiIsInR5cCI6InBhc3Nwb3J0IiwicHB0Ijoic2hha2VuIiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxOTU0NDQ0NzQwOCJdfSwiaWF0IjoxNTkwNjcyNDc2LCJvcmlnIjp7InRuIjoiMjE1OTE0MDQyMSJ9LCJvcmlnaWQiOiIxMjNlNDU2Ny1lODliLTEyZDMtYTQ1Ni00MjY2NTU0NDAwMDAifQ.X7noevZGawXv1Jw1wkaqunTMFVE9FLt7sEX1QSgk0GMJmAHJWnbF5PCdj-Mc7UD2JY_5xvuJU3UlhSvswfK7SQ;info=<
https://cert.example.org/passport.cer>;alg="ES256";ppt="shaken"

With two lines of code in the dialplan, you solve the FCC requirements.
BUT, the caller-ID must be either verifiable associated with the company
that owns Asterisk, or we can supply one for you, from our pool of numbers.
Wireless numbers are not allowed. We check each and call return an error if
the conditions are not met. What happens if you send a random but valid
caller-ID? We still sign it, BUT, with Attestation level "C", which means
we don't know anything about the caller-ID. At some point, carriers will
decline to terminate those calls. It is up to them to terminate or not
those calls.
So what I am doing for the Asterisk community is helping everybody to stay
in business. If you delay the interconnection with me and pretend it is not
urgent, you will end-up in the fauces of nexus, which acts double as a
Certification Authority and Service Provider and charges huge fees. I mean
HUGE.
This wiki should be erased, for it is misleading:

> https://wiki.asterisk.org/wiki/display/AST/STIR+and+SHAKEN

>
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[asterisk-users] STIR-Shaken

2020-05-28 Thread Saint Michael

>
> My company is one if the six service providers approved. We are not ready
> yet, probbably next week, since the certificate needs to be issued by the
> Certification Authority. As I said, we are the ONLY provider that  you may
> use with Asterisk remotely, via UnixODBC. The rest of the other providers
> will force you to send a call to them.

 Here is some material for you to read. Rest assured that this is real.
https://www.fcc.gov/call-authentication
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[asterisk-users] Stir-Shaken for asterisk

2020-05-27 Thread Saint Michael

In a few weeks, no SIP call is going to terminate unless they are signed
properly, as mandated by law.  We are in the business of Stir-Shaken,
signing calls, as an FCC-approved provider. A big differentiator between
our service and the rest: we are the only ones who don't need to receive
the calls in our servers to sign them. We do this over a MySQL call,
easily connectable to Asterisk via res_odbc, so you never have to send us
your calls. This is a sample of how we do this so you may test now:
mysql -u anonymous -h 208.73.232.47 -e "call
strshk.stir_shaken_signature('7274433019','1957408')".
If your caller-ID is a valid US number and not a wireless number (that is a
NO-NO for the FCC), we sign the call as 'C', if you use your own DIDs,
something we can verify as legit, then we sign as 'B', and if you use our
DID as caller ID, we sign as 'A', full attestation.
Please email to venefax at g mail if you have any questions. Do not think
you can do business as usual. The wild west of VOIP is coming to an end.
But we can keep you in business if you follow the rules.
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Re: [asterisk-users] PJSIP sending RTP to private address

2020-05-17 Thread Saint Michael

About this case: the old SIP channel behaves correctly.

On Sun, May 17, 2020 at 2:44 AM Saint Michael  wrote:

> My phone is located behind a NAT, 172.16.0.0/21.
> Asterisk 16 is on a public IP.
> PJSIP has the config below:
> force_rport=yes
> direct_media=yes
> disable_direct_media_on_nat = yes
> direct_media_method=invite
>
> But when I send a call I see the RTP being sent to my private address, vs
> the public IP. This only happens when Asterisk  has dialed the call to
> another carrier. If instead of Dial I choose Answer() and MusicOnHold, then
> the RTP gets shipped to the right address.
> This is a sample of the erroneous behavior:
> Got  RTP packet fromXX.XX.XX.XX:17510 (type 00, seq 024786, ts 017440,
> len 000160)
> Sent RTP packet to  172.16.7.254:50798 (type 00, seq 010736, ts
> 017440, len 000160)
>
> 172.16.7.254 is my private address.
> What am I missing? Should I open a bug?
> Asterisk should never, ever send RTP to a private address when Asterisk
> itself is on a public IP.
> Before you ask, the dialplan is 3 lines,
> '_X.' =>  1. NoOP()
> 2. Dial(PJSIP/${EXTEN}@carrier)
> 3. Hangup()
>
>
>
>
>
>
>
>
>
>
>
>
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[asterisk-users] Help missing

2020-05-16 Thread Saint Michael

I want to see the help when I type core show application , and it's not
available. This is asterisk 16 from sources. I have libxml2-dev  installed.
Ubuntu 19
What am I missing?
Philip
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[asterisk-users] PJSIP does not stop sending invites after call is canceled

2020-05-16 Thread Saint Michael

Endpoint sends an INVITE
Asterisk send an INVITE to the Carrier
Carrier is down, does not even sends ACK
PJSIP sends  several INVITES
End point sends
<--- Received SIP request (397 bytes) from UDP ::50187 --->
CANCEL sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx
:50187;branch=z9hG4bK-524287-1---fbad0437cf02653d;rport
Max-Forwards: 70
To: 
From: "x";tag=a0acbb3e
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
CSeq: 1 CANCEL
User-Agent: Bria 5 release 5.8.3 stamp 102650
Content-Length: 0

PJSIP responds to endpoint

<--- Transmitting SIP response (403 bytes) to UDP:xx:50187 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
X:50187;rport=50187;received=;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "xx" ;tag=a0acbb3e
To: ;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 CANCEL
Server: Asterisk PBX 13.33.0
Content-Length:  0

the PJISP sends an additional response to endpoint
<--- Transmitting SIP response (419 bytes) to UDP::50187 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 172.16.7.254:50187
;rport=50187;received=x;branch=z9hG4bK-524287-1---fbad0437cf02653d
Call-ID: 102650OWFmMWRjMDk0NDUzMzM4MzFhNzcwZDdhZThhMjA1MTk
From: "x" ;tag=a0acbb3e
To: ;tag=5d2fe4a1-b7b1-4868-9696-356511924c60
CSeq: 1 INVITE
Server: Asterisk PBX 13.33.0
Content-Length:  0

to make a long story short, the endpoint sends back an ACK, but after that,
PJSIP keeps sending INVITES to the carrier, which means it did not close
the second leg of the call. If the carrier sends back a 200 OK, there will
be a billing charge, which in case of Mexico is minimum 60 seconds, and the
endpoint will not agree with the charge, resulting in a financial loss for
the Asterisk owner. This is absurd. The second leg must close as soon as a
CANCEL has been received.

The dialplan is only one line
Dial(PJSIP/${EXTEN}@carrier)

Kindly tell me what am interpreting wrong.
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[asterisk-users] New RTP engine

2020-05-11 Thread Saint Michael

>
> Asterisk needs urgently to push the RTP engine to the Kernel, away from
> userland, like professional and commercial softwares do. I measured the
> cost of passing call from a public IP to a private IP, like typically a
> Session Border Controller may do. In Asterisk, ulaw, no transcoding, it
> takes 1.7% of a 3 Ghz core. If the packets where flowing through the
> kernel, like iptables does, it would take 10% if the CPU. Asterisk then
> could be used in hundreds of different roles in the enterprise.  PJSIP has
> no importance at all, this is the big issue. I suggest the developers look
> at an open-source package and adapt the code, is called rtpengine. It uses
> a kernel module to do the job.

Philip
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[asterisk-users] PJSIP crashes

2020-02-26 Thread Saint Michael

>
> I have no control over the SIP calls I receive. PJSIP should log a warting
> and continue. It is causing the CPU usage to spike dramatically.
>
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[asterisk-users] PJSIP crashes

2020-02-25 Thread Saint Michael

PJISP cannot handle the From  field when it does not contain a number.
Can this be fixed?

[Feb 25 12:35:43] ERROR[7143]: pjproject: : sip_transport.c
Error processing 400 bytes packet from UDP 8.38.43.67:5060 : PJSIP syntax
error exception when parsing 'From' header on line 4 col 40:
CANCEL sip:14408785990@162.255.138.102:5060 SIP/2.0
Via: SIP/2.0/UDP 8.38.43.67:5060;branch=z9hG4bK1sansay261086943rdb109274
To: 
From: "Radefeld Dental" ;tag=sansay261086943rdb109274
Call-ID: 1001880886-0-2320154044@8.38.43.49
CSeq: 1 CANCEL
Max-Forwards: 66
Reason: Q.850;cause=34;text="No Ring Timeout"
Content-Length: 0
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[asterisk-users] avoiding any media proxy with PJSIP

2020-02-13 Thread Saint Michael

Is there a guide on how to use PJSIP and never have the media travel inside
Asterisk? No matter what I do, I cannot make this work.
Philip Orleans
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[asterisk-users] How to: send dtmf back to the calling channel from post-answer subroutine executed on outbound channel

2019-12-17 Thread Saint Michael

I have a customer who wants me to send a DTMF on the calling channel if the
called channel says any word. So I am using
[my_gosub_routine]

exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
 same => n,Playback(hello)
 same => n,Return()

[default]

exten => _X.,1,NoOp()
 same =>
n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
 same => n,Hangup()

Is there a way to send DTMF back to the caller from [my_gosub_routine]?
If I use sendDTMF at the moment, it will be heard only by the callee, and
only the caller must hear it.
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Re: [asterisk-users] asterisk-users Digest, Vol 177, Issue 11

2019-05-25 Thread Saint Michael

Joshua
Is there a way in PJSIP to send the audio between the parties always,
unless one of the parties is behind a NAT?
A session refresh would work.
That my only problem with PJSIP. This is routine in the old sip channel.

On Sat, May 25, 2019 at 1:03 PM 
wrote:

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> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
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>
> Today's Topics:
>
>1. Re: Is there a way to make asterisk send a INVITE in-dialog
>   to re-establish the audio (Dan Cropp)
>
>
> --
>
> Message: 1
> Date: Fri, 24 May 2019 17:02:56 +
> From: Dan Cropp 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] Is there a way to make asterisk send a
> INVITE in-dialog to re-establish the audio
> Message-ID:
> 
> Content-Type: text/plain; charset="utf-8"
>
> Thank you Joshua
>
>
> -Original Message-
> From: asterisk-users  On Behalf
> Of Joshua C. Colp
> Sent: Friday, May 24, 2019 9:53 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Is there a way to make asterisk send a
> INVITE in-dialog to re-establish the audio
>
> On Fri, May 24, 2019, at 9:47 AM, Dan Cropp wrote:
> >
> > We are working with an Avaya switch.
> >
> >
> > We send them a REFER. If the transfer is successful, everything is
> > great. If it fails (busy), they send an INVITE in-dialog with a media
> > attribute of inactive. After that, they send a 486 busy.
> >
> > The problem is Avaya basically put the call on hold so audio is not
> active.
> >
> > The Avaya rep is indicating we need to send in dialog invite to get
> > the call audio back? They are essentially saying they put the call on
> > hold because we told them to transfer and it’s our responsibility to
> > take the call off hold.
> >
> >
> > Is there a way to do this?
>
> I don't think there is. We provide the ability in PJSIP to do a session
> refresh[1] but there's no ability to set the stream state like that, so I'm
> not sure what we would specify in that scenario automatically.
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_SEND_SESSION_REFRESH
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
> www.digium.com & www.asterisk.org
>
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[asterisk-users] Compiling error

2019-01-10 Thread Saint Michael

>
> when compiling the latest version, it fails here

./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-asteriskssl
-enable-xmldoc NOISY_BUILD=yes


> gcc -o res_pjsip/config_transport.o -c res_pjsip/config_transport.c -MD
> -MT res_pjsip/config_transport.o -MF .res_pjsip_config_transport.o.d -MP
> -pthread -I/usr/src/asterisk/include-I/usr/include/libxml2  -pipe -Wall
> -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations   -g3  -O3
> -march=native -fPIC -DAST_MODULE=\"res_pjsip\"

res_pjsip/config_transport.c: In function ‘cipher_name_to_id’:

res_pjsip/config_transport.c:982:24: error: ‘PJ_SSL_SOCK_MAX_CIPHERS’
> undeclared (first use in this function)

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c:982:24: note: each undeclared identifier is
> reported only once for each function it appears in

res_pjsip/config_transport.c:982:16: warning: unused variable ‘ciphers’
> [-Wunused-variable]

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c: In function ‘handle_pjsip_list_ciphers’:

res_pjsip/config_transport.c:1106:24: error: ‘PJ_SSL_SOCK_MAX_CIPHERS’
> undeclared (first use in this function)

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

res_pjsip/config_transport.c:1106:16: warning: unused variable ‘ciphers’
> [-Wunused-variable]

  pj_ssl_cipher ciphers[PJ_SSL_SOCK_MAX_CIPHERS];

^

make[1]: *** [res_pjsip/config_transport.o] Error 1

make[1]: Leaving directory `/usr/src/asterisk/res'

make: *** [res] Error 2
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[asterisk-users] Capture SIP all the time

2018-12-05 Thread Saint Michael

Is there a way to configure the old SIP channel to stay in sip set debug
all the time, without human intervention and also at boot time, by default?
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[asterisk-users] jump on DTMF while MP3Player is on

2018-08-18 Thread Saint Michael

>
> I could not find an answer on Google. My MP3Player is busy playing a
> remote radio, and the user presses 7, I want the code to jump to extension
> 7, where there is a different radio. Is this possible?
>


>
>
>
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[asterisk-users] Disable asterisk ssl how to

2018-08-08 Thread Saint Michael

I am trying to install Asterisk 11 on debian 9, and although I use this
parameter in the configure line: --disable-asteriskssl, it goes ahead and
the compilation fails
gcc -o libasteriskssl.o -c libasteriskssl.c -MD -MT libasteriskssl.o -MF
.libasteriskssl.o.d -MP -pthread -I/usr/src/asterisk/include-pipe -Wall
-Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations   -g3  -O3
-U_FORTIFY_SOURCE -D_FORTIFY_SOURCE=2 -march=native -DAST_MODULE=\"core\"
-DAST_IN_CORE
libasteriskssl.c:77:26: error: macro "SSL_library_init" passed 1 arguments,
but takes just 0
 int SSL_library_init(void)
  ^
libasteriskssl.c:78:1: error: expected '=', ',', ';', 'asm' or
'__attribute__' before '{' token
 {
 ^
libasteriskssl.c:87:33: error: macro "SSL_load_error_strings" passed 1
arguments, but takes just 0
 void SSL_load_error_strings(void)
 ^
libasteriskssl.c:88:1: error: expected '=', ',', ';', 'asm' or
'__attribute__' before '{' token
 {
 ^
libasteriskssl.c:97:1: error: expected identifier or '(' before '{' token
 {
 ^
libasteriskssl.c:106:1: error: expected identifier or '(' before '{' token
 {
 ^
libasteriskssl.c:114:27: error: macro "ERR_free_strings" passed 1
arguments, but takes just 0
 void ERR_free_strings(void)
   ^
libasteriskssl.c:115:1: error: expected '=', ',', ';', 'asm' or
'__attribute__' before '{' token
 {
 ^
/usr/src/asterisk/Makefile.rules:143: recipe for target 'libasteriskssl.o'
failed
make[1]: *** [libasteriskssl.o] Error 1

is there a workaround? how do I disable asterisk ssl by editing some file?
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[asterisk-users] How to know the IP of "manager show connected" in dialplan

2018-07-25 Thread Saint Michael

I need to launch a remote process at the machine that has the dialer. I
could
hard-code the IP address in a global variable, but It would be much more
elegant if the dialplan would have a "manager" object where I could read
"manager-->connected". 
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[asterisk-users] How to know the IP of "manager show connected" in dialplan

2018-07-24 Thread Saint Michael

>
> How can I in my dialplan read the IP address that shows in "manager show
> connected". Is this possible?
>
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[asterisk-users] G729 (Dmitry Melekhov)

2018-07-23 Thread Saint Michael

Maybe Digium should include a G729 codec inside Asterisk. What is keeping
them from doing it?

>
> Today's Topics:
>
>1. Re:
> 
> G729 (Dmitry Melekhov)
>
>
> --
>
> Message: 1
> Date: Mon, 23 Jul 2018 08:36:19 +0400
> From: Dmitry Melekhov 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] G729
> Message-ID: <24c4a934-5210-9df4-f9e0-e5681b22b...@belkam.com>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> 20.07.2018 23:35, John Kiniston пишет:
> >
> > On Fri, Jul 20, 2018 at 11:41 AM Saint Michael  > <mailto:vene...@gmail.com>> wrote:
> >
> > The community would benefit if a non/licensed version of G729
> > would be included with Asterisk, since the license expired.
> > The current codec source code posted still requires licensing.
> >
> > I am sure Digium would not prefer to 
> > acknowledge this, but the phenomenal growth of Asterisk is due to
> > the aavailability of a free G729 codec compiled and distributed
> > free by Arkadi Shislov.
> >
> > That'd be a surprise to me with the 325 G.729 licenses I have from
> Digium.
> >
> > I'm not a software pirate, I doubt that most telephony providers are
> > either.
>
> Once again- patent is expired, g729 algorithm is now free.
> You spent you money to wrong place :-)
>
>
>
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[asterisk-users] G729

2018-07-20 Thread Saint Michael

>
> The community would benefit if a non/licensed version of G729 would be
> included with Asterisk, since the license expired. The current codec
> source code posted still requires licensing.
>
I am sure Digium would not prefer to 

acknowledge this, but the phenomenal growth of Asterisk is due to the
aavailability of a free G729 codec compiled and distributed free by Arkadi
Shislov.
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[asterisk-users] AGI fails bad permission

2018-02-23 Thread Saint Michael

Launched AGI Script /var/lib/asterisk/agi-bin/adddnc.php
 adddnc.php: Failed to execute '/var/lib/asterisk/agi-bin/adddnc.php':
Permission denied
The file is of course chmod +x /var/lib/asterisk/agi-bin/*.php
Selinux is disabled
asterisk is running as root with
live_dangerously=yes
in asterisk.conf
The box is Centos 7
What can possibly be causing this?
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[asterisk-users] PJSIP and Non Media Proxy

2017-11-06 Thread Saint Michael

Asterisk is unique in terms that we can create new applications that talk
to databases and generate any logic whatsoever. Asterisk is a development
environment for anything telecom, not a PBX. I believe that we need to make
PJSIP more efficient so Asterisk can expand its footprint.
Please tell somebody to add a way to prohibit PJSIP from proxying RTP. I
can help if you give me some directions, but I understand the complexity of
PJSIP under the hood.
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[asterisk-users] PJSIP and Non Media Proxy

2017-11-05 Thread Saint Michael

Now that Joshua had the kindness to respond, I see here a big disconnect
between Digium and the VOIP industry. 99% of the VOIP entrepreneurs like me
would need to avoid proxying the media.  Would would Digium support and
bring in with such fanfare a channel like PJSIP that lacks the only thing
that 99% would need to do business in an efficient manner? I mean people
like me buy and sale billion of minutes every day, and most of my peers
gravitate towards Opensips and other solution that do not touch the media.
Yesterday I had to roll back my sleeves and go back to the old sip channel.
I would love to see Asterisk-PJSIP to find a way to act like a proxy. This
would turn Asterisk into a real wholesale business tool, which is not, so
far.
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[asterisk-users] PJSIP and Non Media Proxy

2017-11-05 Thread Saint Michael

Please correct me if I am wrong. With PJSIP there is no way for Asterisk to
stay a OUT of the media path, while with the old SIP channel, using
directrtpsetup and directmedia, it just works. The issue I think is that
other servers do not accept reinvites or updates to redirect media, so
PJSIP will not be able to step out ever. Using the old sip channel, the 200
OK with SDP tells the calling side to talk direcly to the other side.
Is there a way to do this with PJSIP?
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[asterisk-users] Broken? SIP/devicename/extension/IPorHost

2017-11-04 Thread Saint Michael

I tried many times and this dial model fails for me
SIP/g729-outbound/155/192.168.1.120
The peer g729-outbound does exist but it does not have a host line,
that is why I am supplying the host dynamically for each call.
According Asterisk13 file configs/samples/sip.conf.sample, this is legal.
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[asterisk-users] PJSIP logging fails

2017-04-12 Thread Saint Michael

I am trying to log my SIP registration attempts.
PJSIP is in logger mode, and I can see INVITES comingh, my SIP Register
does not show, especially the packet I send.
The only thing shown is:
res_pjsip_outbound_registration.c: No response received from 'snet' on
registration attempt to 'sip:7866314772-xnet', retrying in '60'
How do I see my own packet?
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[asterisk-users] Bounty on Google Voice

2017-03-29 Thread Saint Michael

The channel motif and res_xmpp do not work. But there is one company that
does make it work and charges $US 6 for a lifetime connection to your own
free Google Voice number, from SIP. I wonder if anybody would be able to
fix Asterisks libraries so people of low income would not have to pay a
third party for this basic translation service.
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[asterisk-users] Packetization does not work on PJSIP

2017-03-02 Thread Saint Michael

I need to raise my ptime to 60 on my codecs for outbound calls. To that
effect, I add on the endpoint
disallow=all
allow=ulaw:60

and also

 use_avpf   : false
 use_ptime  : true

But the invites always leave with ptime:20.
It used to work fine in the old SIP channel.
What am I doing wrong?
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Re: [asterisk-users] asterisk-users Digest, Vol 151, Issue 23

2017-02-22 Thread Saint Michael

Theory: The carrier is not responding with 100 Trying in the expected time.
Hence, Asterisk is sending the INVITE again.

On Wed, Feb 22, 2017 at 1:00 PM, 
wrote:

> Send asterisk-users mailing list submissions to
> asterisk-users@lists.digium.com
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of asterisk-users digest..."
>
>
> Today's Topics:
>
>1. Looking for Speech Recognition (ASR) suggestions (Dan Cropp)
>2. multiple outbound invites (Jeff LaCoursiere)
>
>
> --
>
> Message: 1
> Date: Wed, 22 Feb 2017 15:43:56 +
> From: Dan Cropp 
> To: "asterisk-users@lists.digium.com"
> 
> Subject: [asterisk-users] Looking for Speech Recognition (ASR)
> suggestions
> Message-ID:
> <41223e927281d842a48cc18032b36cc30118faf...@mail2010c.amtelco.com>
> Content-Type: text/plain; charset="us-ascii"
>
> Is it correct that the unimrcp is the best approach for Asterisk and
> ASR/TTS?
>
> Could anyone provide pros/cons for the various ASR options for Asterisk?
> We need the ability for very large grammars (over 100,000 options).
> Because of this, my initial thought is Nuance or Lumenvox.  Does this sound
> correct?
>
> Have a great day!
>
> Dan
> -- next part --
> An HTML attachment was scrubbed...
> URL:  attachments/20170222/371708a5/attachment-0001.html>
>
> --
>
> Message: 2
> Date: Wed, 22 Feb 2017 11:57:16 -0600
> From: Jeff LaCoursiere 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: [asterisk-users] multiple outbound invites
> Message-ID: 
> Content-Type: text/plain; charset=utf-8; format=flowed
>
>
> Hello,
>
> I have two upstream providers we use for US termination.  The dialplan
> sends calls out the "primary" and if that fails for specific reasons, it
> sends the same call out the "secondary". This has worked well for us
> when we are lazy about keeping balances up, for example.
>
> Starting a few days ago ALL calls sent to the 'primary' were returned as
> busy, though the secondary terminated them fine.  We have a balance, and
> funny enough international calls are going through fine, just not US
> calls.  I opened a ticket.
>
> The response form the carrier is that our asterisk is sending four
> simultaneous invites within one second, and for that reason the call is
> rejected.
>
> I did a packet trace and was able to confirm this is true - only US
> calls sent to this carrier cause our end to send four identical
> simultaneous invites.  When it fails, a single invite for the same call
> is sent to the secondary, which is terminated without issue.
>
> Happy to send the SIP trace if any would care to see it, but is there a
> reason anyone can think of that our asterisk (11.11.0) would suddenly
> start doing this?  It may be that it has been doing it all along, and
> our carrier just started rejected calls that come in this way, I'm not
> sure.
>
> Cheers,
>
> j
>
>
>
>
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> End of asterisk-users Digest, Vol 151, Issue 23
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[asterisk-users] pjsip asterisk 13 add time between DTMF digits

2017-02-05 Thread Saint Michael

I noticed that when I dial some 7 followed by any digit, the other side
gets confused. I would like to double the milliseconds inter-digits in
SendDTMF(). Is there a way to change both the DTMF duration and its
interval?
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[asterisk-users] PJSIP hangupcause how to

2017-02-02 Thread Saint Michael

if a PJSIP call fails, how can I capture SIP code, like 503,603 etc?
in old SIP channel, we had ${HASH(SIP_CAUSE,)}
but in PJSIP it has to be the outbound channel, which is gone when the
control returns to the calling channel.
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[asterisk-users] Still does not work

2016-12-09 Thread Saint Michael

Connected to Asterisk 13.13.1 currently running on siptrunks (pid = 336)
[Dec  9 18:37:34] ERROR[29914]: res_pjsip_sdp_rtp.c:184 create_rtp: Unable
to create RTP instance using RTP engine 'asterisk'
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[asterisk-users] PJSIP missing objects (Saint Michael)

2016-12-03 Thread Saint Michael

There is a mistake on version 13.13.0. I have an application on a
multihomed box, the calls ingress via one IP and egress via the second IP.
It does not work with version 13.13.0.
[Dec  3 18:31:56] DEBUG[188823]: rtp_engine.c:454 ast_rtp_instance_new:
Using engine 'asterisk' for RTP instance '0x7fd0080046e0'
[Dec  3 18:31:56] WARNING[188823]: res_rtp_asterisk.c:2400
create_new_socket: Unable to allocate RTP socket: Address family not
supported by protocol
[Dec  3 18:31:56] WARNING[188823]: res_rtp_asterisk.c:2665 ast_rtp_new:
Failed to create a new socket for RTP instance '0x7fd0080046e0'
[Dec  3 18:31:56] DEBUG[188823]: rtp_engine.c:458 ast_rtp_instance_new:
Engine 'asterisk' failed to setup RTP instance '0x7fd0080046e0'
[Dec  3 18:31:56] DEBUG[188823]: rtp_engine.c:397 instance_destructor:
Destroyed RTP instance '0x7fd0080046e0'
[Dec  3 18:31:56] ERROR[188823]: res_pjsip_sdp_rtp.c:184 create_rtp: Unable
to create RTP instance using RTP engine 'asterisk'
[Dec  3 18:31:56] DEBUG[188823]: res_pjsip_session.c:2494
handle_outgoing_response: Method is INVITE, Response is 488 Not Acceptable
Here

I repeat, if I compile version 13.12.2, ceteris paribus, the app works fine.
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[asterisk-users] PJSIP missing objects

2016-12-02 Thread Saint Michael

In version 13.13.0
 there is no
res_pjsip_keepalive.so
res_pjsip_multihomed.so

Is this normal?
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[asterisk-users] CHANNEL(accountcode) only stores 19 chars

2016-11-11 Thread Saint Michael

I cannot store more than 19 chars in CHANNEL(accountcode). I can certainly
store more in the code, but my CDR table, which has an accountcode field
varchar(256) only shows 19. There is a disconnect somewhere.
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[asterisk-users] PJSIP how to change the generated SIP CALL ID

2016-10-17 Thread Saint Michael

I need to change the automatically generated SIP call ID, from (example)
64f61c7a-c68f-498b-8661-b42e5c771363
to
64f61c7a-c68f-498b-8661-b42e5c771...@my.ip.add.ress

since that is the only way to make sure the calls came from my box.
How do configure this in the system? It should be user-configurable.
Philip

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[asterisk-users] Installing Asterisk on MAC native

2016-09-20 Thread Saint Michael

I need to install Asterisk on a MAC, native, no virtualization.
Has anybody done this? Are there documents on the Internet?
I googled it and all web sites that claimed to help installing Asterisk on
a MAC have disappeared. Is it possible at all?
Digium should actually have a MAC app in the Apple store with a PBX. It
should be a paid app. I would buy it right away.


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[asterisk-users] Originate Fails inside Dial() M(macro) or U(subroutine)

2016-09-12 Thread Saint Michael

Is this by design? I execute  a Dial() with a parameter M(macro) or
U(subroutine). Inside either I use the application originate. The variable
${ORIGINATE_STATUS} says "SUCCESS" but no call is actually executed. Should
I file a bug report?
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[asterisk-users] No ringback heard

2016-08-25 Thread Saint Michael

I dial two destination like this

Dial(PJSIP/endpoint1/sip:${EXTEN}@${IPA}/endpoint1/sip:${EXTEN}@
${IPB})

But I need the audio from one of them to be heard by the caller.
None gets heard. I switch the order but nothing.
How I get the audio for  one in particular?
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[asterisk-users] Dial and start music on hold after timeout

2016-08-24 Thread Saint Michael

I have the same exact issue. I cannot push any sounds or even Playtones to
the caller, unless the channel is answered, which is not possible for
billing reasons.
I am also using the Local channel & Dial(PJSIP/...).
I think this is a bug in Asterisk 13. The Dial function has not answered
yet, so the Local channel should be able to play anything to the caller,
without answering, in parallel with Dial.
Should I open a JIRA ticket?
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[asterisk-users] Fwd: Backport fix

2016-08-17 Thread Saint Michael

There is a big bug
https://issues.asterisk.org/jira/browse/ASTERISK-24768
It affects version 11, but it was fixed only from 13.20 onwards.
However, millions of people still use version 11. This bug makes Asterisk
crash every few hours under any load that has RTP going through Asterisk.
For example, after
3 hours and 6 minutes
 lsof | grep asterisk | grep FIFO | wc -l
245122
with less than 120 calls, with fill media proxy.
and it crashes
Can somebody help?
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[asterisk-users] PJSIP is Ignored

2016-08-12 Thread Saint Michael

Asterisk 13.11 rc1

./configure   LDFLAGS="-z muldefs" --libdir=/usr/lib64
--with-unixodbc=$(odbc_config --include-prefix)/ --disable-dev-mode
--with-pjproject-bundled

checking for pjsip_dlg_create_uas_and_inc_lock in -lpjsip... no
checking for pjsip_tsx_create_uac2 in -lpjsip... no
checking if "pjmedia_mod_offer_flag flag =
PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE" compiles using pjmedia.h... no
checking for pjsip_get_dest_info in -lpjsip... no
checking for pjsip/include/pjsip/sip_util.h in -lpj... no
checking for pjsip_endpt_set_ext_resolver in -lpjsip... no
checking if "struct pjsip_tls_setting setting; int proto; proto =
setting.proto;" compiles using pjsip.h... no
checking for pjsip_evsub_add_ref in -lpjsip... no

When I do make menuselect, it is disabled.

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[asterisk-users] PJSIP not detected

2016-08-11 Thread Saint Michael

I installed PJSIP from the project
git clone https://github.com/asterisk/pjproject pjproject
cd pjproject
make uninstall & make distclean
./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound
--disable-resample --disable-video --disable-opencore-amr
--with-external-srtp
make dep && make && make install && ldconfig && ldconfig -p | grep pj

and it is there, but the configure for Asterisk 13.11.0-rc1 does not detect
it and it cannot compile it.
What am I doing wrong? The box is Ubuntu 14.04 LTS
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[asterisk-users] Detecting end of Ringback

2016-08-01 Thread Saint Michael

On an outbound call, PJSIP, I execute a macro and try to detect the end of
ringback, inside the macro. So far waitforring() does nothing, stays stuck.
Any combination of waitffornoise and waitforsilence, or backgrounddetect
fail to find the moment when the ringabck stops. I can detect when it
starts, with waitfornoise, but not its end.
Any idea? Should I open a ticket for waitforring?
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[asterisk-users] PJSIP and the pound (#) as %23

2016-07-20 Thread Saint Michael

Is there any way to make PJSIP send the "#" as "#" and not as %23 in the
INVITE?
I cannot figure this out.
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[asterisk-users] ODBC freezing Asterisk 13

2016-07-14 Thread Saint Michael

Many people are reporting the same issue, so it is not my imagination.
Asterisk 13 above 13.1 is useless for anybody who relies on res_odbc.so.
As you know, after that version, the dropped the complexity of Pooling onto
unix_odbc itself. Not so simple, it seems. I noticed that after a few hours
of inactivity, any call to func_odbc-defined funcions will block and hang
for ever. All we can do at that point is reset Asterisk.
I think it was highly rushed a decision to drop all the work done in ODBC
inside Asterisk. Maybe unix_odbc pooling is not ready, has bugs, it cannot
be used in production. I don't know what the issue is, but I had to
downgrade to Asterisk 13.1 and my ODBC problems disappeared. Asterisk did
not need to drop the ODBC pooling code. It did work. It should be fixed,
made faster, etc.
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[asterisk-users] Blocking 180 Ringing with PJSIP How to

2016-06-27 Thread Saint Michael

I have a business need to block any 180 Ringing packets coming from an FXO
gateway, Grandstream 4108. I use Asterisk 13.9.1, and PJSIP. Is this even
possible? I all I may do is hack the source code,  will it be PJSIP code or
Asterisk code? Any help on where to look for the relevant code?
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[asterisk-users] Possible BUG on Asterisk 13.9.1

2016-06-26 Thread Saint Michael

I get this message very often:
[Jun 26 23:52:49] ERROR[10396]: channel.c:1278 ast_channel_by_name_cb: BUG!
Must supply a channel name or partial name to match!
I could file a bug if somebody tells me how to obtain a trace for this. I
cannot imagine.
I use PJSIP
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[asterisk-users] update from Asterisk 12 to 13

2016-06-21 Thread Saint Michael

I am using Asterisk 12 and PJSIP. Last night I tried to upgrade to
Asterisk 13, and it did compile just fine, but PJSIP would not load, and no
error was shown on the screen when I did "pjsip reload".
Do I have to erase some objects before compiling Asterisk 13? Is there a
document that shows steps to a successful migration?



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[asterisk-users] Execute an app on the master channel from inside a Macro on the called channel

2016-05-03 Thread Saint Michael

While I am executing a Macro on the called channel, right after the call
connects, I need to execute an app on the master channel, from inside that
macro, specifically, SendDTMF. If I execute it now, it send a text message
to the Callee, when my app needs to send it to the caller.

I could use
set(master_channel(variable)=XXX), but then how do I execute some code on
the master channel.
Note that I could send the name of the master channels to the Macro
M(Name^parameter), but then how do I execute SendDtmf on the identified
Master Channel?
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[asterisk-users] Re: Recommendations for free virtual server tech and Asterisk? (Ikka Tirtawidjaja)

2016-04-09 Thread Saint Michael

OpenVZ is useless for Asterisk or any other resource intensive
application. OpenVZ was built from a hosting provider point if view, and if
you exceed any of the counters, dozens of them, they system will
kill your app immediately. It is almost impossible to build a VPS that will
use all the resources of the machine.  The only container technology that
works for Asterisk is LXC, better implemented by Ubuntu on the server side.
Centos 7 is behind in the LXC version and it is part of the core OS, but
found in a repository. Yo need kernel 4.X to make it works flawlessly.

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[asterisk-users] Subject: Re: Incoming INVITE with Portability Info

2016-03-20 Thread Saint Michael

My company http://prescott-clearwater.com/ is the largest provider of LRN
dips for Asterisk, via ODBC. I will be happy to help you dip any number,
from a few to 50 million per day.

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Re: [asterisk-users] asterisk-users Digest, Vol 140, Issue 15

2016-03-12 Thread Saint Michael

It does not work. That was the first think I tried.
Maybe we need a patch?
I don't want to file a bug if there is a workaround.

On Sat, Mar 12, 2016 at 1:00 PM, <asterisk-users-requ...@lists.digium.com>
wrote:

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> Today's Topics:
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>    1. what to do when a sip password includes a semicolon
>   (Saint Michael)
>2. Re: what to do when a sip password includes a semicolon
>   (Kevin Larsen)
>
>
> ----------
>
> Message: 1
> Date: Fri, 11 Mar 2016 14:43:47 -0500
> From: Saint Michael <vene...@gmail.com>
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users@lists.digium.com>
> Subject: [asterisk-users] what to do when a sip password includes a
> semicolon
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[asterisk-users] what to do when a sip password includes a semicolon

2016-03-11 Thread Saint Michael

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.
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[asterisk-users] How to execute a macro after dial but before connect

2016-02-19 Thread Saint Michael

Dear friends:
Is there a way to execute a macro or sub-routine after we send the invite
before we receive anything like a 200 OK, 183, etc?
Philip
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