Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

[James Cloos]

>>>>> "JC" == Joshua C Colp  writes:

JC> To be specific, this is in PJSIP land. There was no insisting or anything
JC> and it wasn't a decision we originally made. It's the way that Teluu
JC> implemented the TLS transport in PJSIP and since we use PJSIP then it
JC> applies to us.

my recall is more likely a bit older than that, before pjsip.

there was a thread either in bugs or on one of the lists.

but as later notes pointed out (and i really ought to have thought of ☹)
it is only relevant, as you noted, if verify is on.

at the time i was a fan on wildcards.

then le came along, and then added dns01 support.

now i prefer a separate cert each plus a 3/1/1 tlsa for each port.

but at the time it was anoying.

-JimC
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Re: [asterisk-users] PJSIP to Twilio over TLS - wildcard cert problem

[James Cloos]

>>>>> "KT" == Kingsley Tart  writes:

KT> I can't get Asterisk to send a SIP call to Twilio over TLS
KT> because it complains about Twilio's wildcard certificate.

the sip rfc claims that wildcard certs should be invalid for sip.

digium insisted on following that advise as set in stone, and so
asterisk refuses such certs.  i doubt that stance is different
under sangoma.

the only workaround is to remind twil of the rfc and get them to
replace the wildcard with an rfc-copliant cert.  at least for the
sip ports.

-JimC
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[asterisk-users] ?????? Does Asterisk-16.1.1 support "make freepbx"

[james]

thanks. I also tried that, failed. 




--  --
??: "Jorge Mart??nez L??pez";
: 2018??12??31??(??) 5:26
??: "Asterisk Users Mailing List - Non-Commercial 
Discussion";

: Re: [asterisk-users] Does Asterisk-16.1.1 support "make freepbx"



On Sun, 30 Dec 2018 at 15:10, Joshua C. Colp  wrote:
>
> On Sun, Dec 30, 2018, at 9:03 AM, james wrote:
> > hello:
> > Does asterisk-16.1.1 support freepbx by default?
>
> No version of Asterisk currently has any built in mechanism to install and 
> set up FreePBX. They operate as separate projects and the FreePBX install 
> instructions would need to be used to install it.
>

The article in the original post was published on December 28th when
Spain celebrates their April's Fools.

Greetings,
Jorge

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[asterisk-users] Does Asterisk-16.1.1 support "make freepbx"

[james]

hello:
Does asterisk-16.1.1 support freepbx by default?

/contrib/scripts/install_prereq install

./configure

make

make install

make config

make freepbx // I run this command without success. 

after then, we can access the freepbx GUI? 

this website show it work:

https://www.sinologic.net/2018-12/la-ultima-version-de-asterisk-16-incluye-un-comando-para-instalar-freepbx.html-- 
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Re: [asterisk-users] T-38 re-invite issue

[James Cloos]

>>>>> D'Arcy Cain  writes:

>> Ie after both sides select t38, until they agree on the t38 terms.

> OK, so does that mean that setting it to 25000 should leave time for the
> re-invite or does the timeout start after that.

As I wrote above, after that.  After the sip/sdp.

-JimC
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Re: [asterisk-users] T-38 re-invite issue

[James Cloos]

>>>>> "DC" == D'Arcy Cain  writes:

DC> Perhaps someone can explain what t38timeout is supposed to do

A 'git grep t38timeout' on the src leads one to res/res_fax.c, where one
case see that it is the number of miliseconds to permit for t38 negotiation
to complete once it starts.

Ie after both sides select t38, until they agree on the t38 terms.

-JimC
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[James B. Byrne]

On Fri, June 16, 2017 12:28, Tim S wrote:

Whether it is intentional or not these messages railing against the
list operators has a decided tone of condescension which is not
warranted.  The fact of the matter is that DMARC is broken by design
and the unpleasant effects that adoption of it has on mailing-list
traffic were well hashed out on the ITEF mailing lists before it was
adopted anyway.  What was predicted there has come to pass.

DMARC conflicts with the existing SMTP RFCs in several ways, none of
which I will elaborate here but all of which may be discovered by
perusing the relevant threads on the ITEF mailing lists.  Some mailing
list management software, notably Mailman, since has been modified to
'work around' the problems with DMARC if so configured by the list
owners.  But only at the cost of violating the SMTP RFCs themselves.
Do not take my word for it.  Raise these issues on the Postfix mailing
list and discover what response you get from Viktor and Wietse.

The driving force behind DMARC was YAHOO's shoddy security of their
own users' accounts.  With Hotmail and similar ilk close behind. It is
a completely inappropriate, and in my opinion ill-thought-out,
technical solution to what is essentially an internal security problem
at some email providers, albeit very large ones.  In general it is an
example of what is called 'externalising your costs'.

The appropriate answer has been provided: lose the
gmail/hotmail/yahoo/freemail account and administer your own domain
for personal email. Configure the spf and dkim settings on your own
domain as required to suit your needs and not those of someone else.

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[asterisk-users] Snom870 FW:8.7.5.35

[James B. Byrne]

Does anyone on this list know how to make the Snom870 with FW:8.7.5.35
display the Caller ID in the display field while the ringing either
together with, or instead of, the topmost virtual key in the info
column?  I realise that the purpose of having the virtual key display
the caller ID so as to allow selection of which incoming call to take.
 But the resulting display size is so small as to make that
information unusable.


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Re: [asterisk-users] OT: DMARC enabled domains on this list

[James B. Byrne]

On Mon, June 5, 2017 15:30, Daniel Tryba wrote:

>
> The reports are there to tell you something isn't right (like on this
> mailing list). Disabling them is only hiding the problem, people might
> be replying with the correct answer to a problem, but the OP might
> never gets that message.
>

What DMARC reports is that somebody other than yourself is sending
email claiming to be you.  And there is absolutely nothing that you
can do about it.  So the question arises: What is the value in these
reports?


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Re: [asterisk-users] OT: DMARC enabled domains on this list

[James B. Byrne]

On Fri, June 2, 2017 16:30, Doug Lytle wrote:


This is likely the issue surrounding mailing lists rewriting headers
and/or modifying messages bodies or simply re-transmitting messages as
the original sender from an unapproved domain. This was discussed at
length on the ITEF mailing list.  Without seeing your headers and
those of a recipient it is impossible to be sure but my spidy sense
tells me this is so.

You can manage this in your DNS forward zone by turning off the DMARC
reporting request. No, I no longer recall the details.  Or you can
simply direct the incoming reports to /dev/null.

As I get the digest version of the list the message sender and domain
match DMARC provisions, if any are set for digium.com.

HTH.


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Re: [asterisk-users] Let's encrypt privkey : Specified certificate file could not be used

[James Cloos]

>>>>> "JK" == Jonas Kellens <jonas.kell...@telenet.be> writes:

JK> [Jun  2 14:29:28] ERROR[27360][C-0ae5]: res_rtp_asterisk.c:1441
JK> ast_rtp_dtls_set_configuration: Specified certificate file
JK> '/etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' for RTP instance
JK> '0x7f920c538a78' could not be used

That error means that openssl's SSL_CTX_use_certificate_file() returned
an error.

The later error is just a result of that one.

Does the uid/gid used for asterisk have access to the key?

If the uid you use for asterisk is called asterisk, run this as root:

su -c 'cat /etc/letsencrypt/live/ws.mydomain.tld/privkey.pem' - asterisk

If it fails, then the problem is permissions.

You may need to alter the permissions on /etc/letsencrypt to allow
non-root uids to access the symlinks and their targets.

-JimC
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[asterisk-users] SNOM870 provisioning BLF settings

[James B. Byrne]

We use Snom870s together with Asterisk 13.14.0 and FreePBX
13.0.191.11.  I am having an ongoing problem with setting the BLF
values on these phones from the configuration file generated from
FreePBX.

In FreePBX we employ the Commercial Endpoint Manager (CEM) to
configure these phones.  The resulting configuration file contains
lines like these:

blf sip:10@;
blf sip:11@;
blf sip:12@;

However, on the test phone I am using (FW 8.7.5-35) I see this as a
result:

Context TypeNumber  Short Text
Active  LineP1
Active  LineP2
. . .
Active  LineP15

When instead I have expect to see:

Context TypeNumber  Short Text
Active  BLF <sip:10@addr;user=phone>|** Recep   P1
Active  BLF <sip:11@addr;user=phone>|** AKL P2
. . .
Active  BLF <sip:24@addr;user=phone>|** JillP15

I have rebooted this phone several times and the provisioning web
server records the transfer of the settings file with a 200 result.

"GET /snom870.htm HTTP/1.1" 200 224 "-" "Mozilla/4.0 (compatible;
snom870-SIP 8.7.5.35 SPEAr300 SNOM 1.4 000413419A8A)"

But the settings file appears to have no effect on the BLF
configuration.  I am at a loss at this point.

Any suggestions?

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Re: [asterisk-users] iaxModem pickup problem

[James B. Byrne]

On Thu, May 4, 2017 13:19, Telium Technical Support wrote:
. . .
> This design (FreePBX) makes Asterisk much more fragile than it has to
> be.
> It's a good idea to keep a backup astdb on the PBX in case of
> corruption.
>

I have added a cron job to make a copy that file every day at midnight
with a date timestamp in the file name.  I also have daily scheduled
backups of the entire FreePBX installation and databases through
FreePBX itself but this approach seems a little more convenient.

We have had similar incidents in the past which we could never
determine the cause of before it was somehow rectified.  I infer that
on each occasion something we tried simply caused the astdb file to be
rebuilt and thereby corrected the issue without us ever being aware
that is what happened.

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Re: [asterisk-users] iaxModem pickup problem

[James B. Byrne]

On Thu, May 4, 2017 11:38, Telium Technical Support wrote:
> It depends a bit on your version of FreePBX, but here's a link to show
> you how:
>
> http://telium.ca/pages/forums/viewtopic.php?f=7=19
>
> Hopefully option 1 works for you (quick and easy).  If not, you'll
> have to try option 2.  Ignore option 3 since that's only for users
> of High Availability for Asterisk (HAAst).
>
> (I assume that if you had a full backup you would have already tried
> to restore it)
>

No, I did not try to restore from backups; and yes I have daily
backups to recover from if that is necessary.  However, I have since
corrected the damaged rows in astdb.sqlite and the fax service is now
working again.

If someone could explain what likely happens to damage astdb.sqlite
when the system is suddenly powered off I would appreciate it.

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Re: [asterisk-users] iaxModem pickup problem

[James B. Byrne]

On Thu, May 4, 2017 10:22, James B. Byrne wrote:

I am advised that it may be possible thast the astdb.sqlite3 database
may be corrupted.  Are there procedures to rebuild or repair this? 
Where are they documented?  If not then how does one repair such?

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[asterisk-users] iaxModem pickup problem

[James B. Byrne]

We run Asterisk 13 using the FreePBX 13.0.190.19 distro based on
CentOS-6.4.  We also run HylFAX+ 5.5.3 with iaxModem 1.2.0 on the same
system with AdvantFAX as the web front-end.  Our two fax lines are
configured as iax2 DEVICES.

These components have been working together through various versions
since 2013.  On Tuesday last our site was subjected to a prolonged
power outage that drained our twin UPS set up flat resulting in a
power down state for the Asterisk host.

Upon power recovery the asterisk host system came up, the phone system
works, but we are now unable to receive faxes through Asterisk.  We
can send faxes but not receive them.  The fax line never picks up and
a redirect to a voice recording informing voice callers of their
mistake is triggered instead.

I have a trace of the asterisk 'full' log that captures one of these
failed calls and I would like some help in determining if a clue to
what has happened is contained therein.  I do not wish to simply post
it to the list with getting permission since it is quiet long.  But I
do need to get this resolved and I cannot fathom why the lines are not
picking up incoming faxes.

Any help would be gratefully accepted.


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

[James Cloos]

>>>>> "FM" == Fabio Moretti <fmore...@tecytal.com> writes:

FM> when a call enter, asterisk sense it and store its values (callerid,
FM> date and time, etc) somewhere, but nothing more, people will answer
FM> using the old analog phone.  The goal is to have a log of the inbound
FM> calls without touching the old analog system (it's shared between
FM> different subjects).

IIUC, the pots line has both some number of analog phones a/o fax
machines on it, plus a fxo->sip gateway, yes?

You can route the sip portion to asterisk and have the dialplan log
everything but never answer.

You may want to call the Ringing dialplan application, but even that
may not be required.  OTOH, calling Ringing should prevent the gateway
from assuming that the asterisk machine never saw the INVITE.

Eventually, when the other extension answers, the fxo->sip gateway will
cancel the sip call just like it would if the caller hung up.

(There is a possibility that any given gateway may not cancel the sip
call until the analog call is completed; you need to test.)

-JimC
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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

>>>>> "DC" == D'Arcy Cain <da...@vybenetworks.com> writes:

DC> I did debug 10 and saved the console output into files which I
DC> compared side by side.  No material difference.

In that case I'd add more debug statements to apps/app_voicemail.c (in
vm_exec()), including a log at the start of what is in *data and args.

Looking at it, it only plays vm-whichbox when ast_strlen_zero(data),
which implies that the args to Voicemail are not making it through.

-JimC
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Re: [asterisk-users] Voicemail asking for login

[James Cloos]

I enable full log and run 'core set debug 9' before doing a pair of
tests.

(The full log is easier to grep than the console output.)

Then compare a working vs stocktrans2 side by side.

-JimC
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Re: [asterisk-users] E-911

[James Cloos]

There are a few services available, in addition to the offerings from
(some? most?) upstreams.

I'm aware of http://www.bulkvs.com/e911.html

It is probably the most affordable option if your upstream does not
offer it.

I use their origination and termination services, but have not needed to
try their e911.  They only charge $0.72/month/number for e911.

(The way e911 works each number which may be used as callerid for 911
calls must have an address added to the e911 database.  You then need
to arrange to use a/the number corresponding to the address where the
emergency is as the userpart in the SIP From: header.)

-JimC
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Re: [asterisk-users] inbound T38 to email

[James Cloos]

>>>>> "JL" == Jeff LaCoursiere <j...@jeff.net> writes:

JL> Is there any new modern way to take t38 from a (SIP) DID provider and
JL> route to email?  Thanks for any insight :)

With recent versions of asterisk you can use the ReceiveFax and SendFax
dialplan applications.  They can be configured (in res_fax.conf) to use
t38 when available.

-JimC
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Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

[James Cloos]

I saw this on the bug list first and sent a reply, but for the archives
I'll copy it here, too.

REMAINDER() calls libm's remainder(3) or remainderl(3), infix % calls
fmod(3) or fmodl(3).

remainder(3) is defined to round the quotient to the nearest int (always
using round-to-even, notsithstanding the rounding mode), fmod(3) is
defined to trunc(3)ate the quotient.

So the result of x%y will always be in the range [0,x] and the results
of remainder(x,y) will be in the range (-y/2,y/2].

-JimC
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Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

[James Thomas]

All I can tell you is where -3 comes from.
>From http://www.voip-info.org/wiki/view/Asterisk+Expressions :
REMAINDER(x,y) computes the remainder of dividing x by y. The return value
is x - n*y, where n is the value x/y, rounded to the nearest integer. If
this quotient is 1/2, it is rounded to the nearest even number.

-3 comes from:
n = x/y = 957/60 = 15.95 which rounds to 16
n*y = 16*60 = 960
x - 960 = 957-960 = -3

I'm not mathematically gifted either but I think the n is the problem. it
shouldn't be the rounded result it should be the integer part of x/y (n=15)

Can you just use modulo instead: ${MATH(${myNum}%60,int)}
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Re: [asterisk-users] pbx testsuite

[James Wystead]

I'd like to see it.
That definitely interests me.

Thanks Glen
On Jun 15, 2016 9:05 AM, "Marek Červenka"  wrote:

> hi,
>
> we have in house developed pbx testsuite based on
>
>- node.js
>- selenium
>- protractor
>- gulp
>- pjsip - pjsua python
>- docker
>
>  there are helpers for testing
>
>- sip
>- web
>- api
>
> you can create end-to-end scenarios like
> - create 2 users via web
> - call from first user to second
> - check CDR result via API
>
> but
> we have some problems in "burning" tests with frozen jasmine reporter,
> with account management in pjsua python, ...
>
> my questions are:
> is there some similiar testsuite based on node.js technology? (i know
> about asterisk-testsuite and xivo-testsuite)
> is there interest in publishing our testsuite on github?
>
> --
> ---
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> ===
>
>
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Re: [asterisk-users] Windstream SIP Trunk settings

[James Cass]

Here's what I ultimately got to work (in case it helps someone):

Name your trunk
Enter your outgoing CID

Under Outgoing settings-
Trunk name - whatever you choose to name it
PEER Details-
host=IP address of SIP gateway
type=friend
context=from-trunk
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite

Incoming settings -

none

Registration string -
username:passw...@xxx.xxx.xxx.xxx

James Cass <http://goog_987864563>
jcas...@gmail.com


On Tue, Feb 23, 2016 at 12:20 PM, Rodrigo Ramírez Norambuena <
decipher...@gmail.com> wrote:

> February 23 2016 9:37 AM, "James Cass" <jcas...@gmail.com> wrote:
> > Thanks everyone, all sound advice. Still can't even get the calls to
> show up on the console at all
> > - I suspect the issue is on the WS side, as I'm not having any issues
> with other carriers with
> > similar settings.
>
> You can debug SIP to detect the problem. May be exists some cause tell you
> more information in the
> trace SIP.
> --
> Rodrigo Ramírez Norambuena
> http://www.rodrigoramirez.com
>
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Re: [asterisk-users] Windstream SIP Trunk settings

[James Cass]

Thanks everyone, all sound advice.  Still can't even get the calls to show
up on the console at all - I suspect the issue is on the WS side, as I'm
not having any issues with other carriers with similar settings.

Thanks again.

James Cass <http://goog_987864563>
jcas...@gmail.com


On Mon, Feb 22, 2016 at 9:06 AM, Mark Wiater <mark.wia...@greybeam.com>
wrote:

> In my case, username is the BTN. I also set the fromdomain to be the sbc
> that I'm registering with. Externip might help also?
>
> [paetec]
> host=10.250.0.5
> username=btn
> fromdomain=10.250.0.5
> dtmfmode=rfc2833
> externip=10.255.0.2
>
> I've used these settings on both registering and non-registering trunks,
> connecting to both the Broadworks and Plexus platforms in Windstream.
> Though all of my asterisk versions have been 1.8.x
>
> Mark
>
>
> On 2/22/2016 8:20 AM, James Cass wrote:
>
> Does anyone on this list use Windstream as a SIP trunk provider?
>
> If so, would you mind sharing your peer settings?
>
> I'm using asterisk 13.7.2 and can't seem to get the inbound working
> correctly (using registration).  Outbound is fine, but they are seeing an
> authentication error on their end.
>
> Here are my inbound peer settings:
>
> username=
> secret=
> host=
> type=peer
> fromuser=
> context=from-trunk
> dtmfmode=auto
> canreinvite=no
> qualify=yes
> insecure=port,invite
>
> register string:   :@:5060
>
> Thanks in advance,
>
>
>
>
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[asterisk-users] Windstream SIP Trunk settings

[James Cass]

Does anyone on this list use Windstream as a SIP trunk provider?

If so, would you mind sharing your peer settings?

I'm using asterisk 13.7.2 and can't seem to get the inbound working
correctly (using registration).  Outbound is fine, but they are seeing an
authentication error on their end.

Here are my inbound peer settings:

username=
secret=
host=
type=peer
fromuser=
context=from-trunk
dtmfmode=auto
canreinvite=no
qualify=yes
insecure=port,invite

register string:   :@:5060

Thanks in advance,
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[asterisk-users] Asterisk & Docker

[James McDonald]

Given that SIP packets have embedded IP addresses in them and when behind
NAT they need to have IP settings to deal with it.

How does one set up Asterisk when it is in a docker container?

E.g.

Docker Container IP <==> Docker Host <==> Gateway <==> Internet

172.17.0.1/16 <==> 192.168.50.1 <==> 192.168.50.254 <==> External Internet
IP

It seems to me that we are doing double NAT?


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Re: [asterisk-users] PJSIP Stun/ICE

[James Cloos]

>>>>> "AS" == A J Stiles <asterisk_l...@earthshod.co.uk> writes:

AS> If you are paying for a business-grade Internet connection, you
AS> should get a static IP address -- or a block of them -- as
AS> standard.  Maybe you need to change your ISP?

In some places (including here) static ip is not affordable.

-JimC
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Re: [asterisk-users] PJSIP Stun/ICE

[James Cloos]

>>>>> "JC" == Joshua Colp <jc...@digium.com> writes:

JC> I disagree that it makes it worthless for a large number of
JC> users. It's only within the last few days that a few people have run
JC> into this particular issue where they have a public IP address that is
JC> changing a lot and PJSIP does not support changing it without a
JC> restart. If it were a huge sweeping issue we'd be seeing it more
JC> often. If it continues to show up a community member or us (heck maybe
JC> even myself in my spare time) may look into implementing it.

It is only in the last few days that this discussion occurred.  This is
not the first mention of problems with using pjsip on dynamic ips.

Most affected users are probably still using chan_sip.  Or haven't even
upgraded to 13 yet.

I gave up switching my edge asterisk to pjsip at least twice because I
couldn't figure out how to configure it properly for a dynamic ip.  And
I sent a note to one of the lists at least on the 2nd attempt.

That install doesn't need nat for sip/rtp since it runs on the router,
but it does need to handle dynamic ip.

In short, this breaks sip for nearly everyone using asterisk at home and
even a lot of businesses.

It may not break it every day, but it is enough to drive a lot of people
away from asterisk once they learn of it.

JC> The support level for chan_sip has already been changed and was
JC> announced long ago.

had this issue been noted in that announce you'd have received -- I
expect -- quite a few complaints.

This flies in the face of all of the (very welcome) work which went into
supporting reload rather than restart.

Getting pjsip to support changes on a reload would be an acceptable
first step.

-JimC
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Re: [asterisk-users] PJSIP Stun/ICE

[James Cloos]

>>>>> "JC" == Joshua Colp <jc...@digium.com> writes:

JC> This stems from PJSIP not being dynamic with transports (it
JC> doesn't like its environment changed to that degree while
JC> in use). I'm afraid if your IP changes you'd have to restart
JC> Asterisk when you are using PJSIP.

Wow.

I say this having voted for pjsip over the listed alternatives back when
the plan to depricate chan_sip was first floated:

That should have excluded pj from the options.  Which of course means
there were no reasonable options.

Can ari get around that bug? 

Lack of full support for traversing nat makes pjsip worthless for a
large number of users.  And the whole point of realtime is to have all
of the rt config fully dymanic.

If ari cannot avoid that limitation, chan_sip should get full ongoing
maintainance until pjsip is fixed.

-JimC
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[asterisk-users] ARI all subscribe

[James Cloos]

I wasn't able to make it back to the devcon after lunch or to as many of
the talks as I'd have liked (the excessive a/c exacerbated by symptoms
enough to be painful), so I probably missed something relevant to
this...

What is the syntax of an ALL subscription websocket url in ari?

I'd like to use wscat with such a sub to get a better idea of what the
various events look like.

Thanks,

-JimC
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Re: [asterisk-users] asterisk server stress test

[James Cass]

Steve, would you be willing to share that quick bash script?

James Cass http://goog_987864563
jcas...@gmail.com


On Wed, Aug 19, 2015 at 12:11 PM, Steve Edwards asterisk@sedwards.com
wrote:

 On Wed, 19 Aug 2015, Dominique Haeber wrote:

 Hi Barry Flanagan,

 Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:

 SIPP is probably what you seek. http://sipp.sourceforge.net/

 Hope this helps.


 That looks pretty like what I'm looking for! Many thanks!


 Another approach is to use another Asterisk system.

 Recently, a customer wanted to confirm his platform would support 500
 simultaneous calls.

 I wrote a quick bash script to dump 500 call files (at a leisurely pace)
 into another host that originated calls to the target host.

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST


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Re: [asterisk-users] How to enable IM over the asterisk server

[James Cass]

You can have the openfire server installed on the same server as asterisk
without any issue, just size your server appropriately.  Just keep in mind
they are different services.

James Cass http://goog_987864563
jcas...@gmail.com


On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote:

 I just get started with it so my question maybe not well catch. Anyway to
 do the VOIP call and IM we need to use two difference servers? which one is
 asterisk for VOIP ? and other one for IM that is openfire ? or we can have
 other choice better than this ?
 Thank you for your help, I am waiting for your reply.

 Thyda


 On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Hi Thyda,

 I think you should see these as two individual systems. (I'm not an
 expert so just thinking out loud).

 Since you mention that you did a SIP mapping on Openfire, may I assume
 that you have the Asterisk IM plugin?

 In case of yes:
 Yes, there is a plugin between OpenFire and Asterisk but it is not
 actively developed anymore since 2006
 http://www.igniterealtime.org/projects/asterisk/

 So I don't think the plugin is really realiable anymore  on current
 versions.

 --

 I consider them as 2 separate systems which have to work on their own.
 Unfortunatly this means that every softphone has 2 accounts: one is SIP to
 Asterisk, one is XMPP to Openfire.

 That way our users are able to call internal/external using Asterisk, but
 do IM and internal calling via Openfire. (They can choose which source they
 take)

 Openfire is connected to our AD so our users just can logon with their
 Windows credentials.

 Unfortunatly, if you want a real production connection between Asterisk
 and Openfire, I'm unable to assist since I don't have the knowledge of it.
 sorry

 Hope this helps a bit.
 kristof
  Thyda ENG ength...@gmail.com 7/07/2015 11:28 
 Actually, I am using the openfire and I create two users with the SIP
 mapping on the openfire to the asterisk server. I can register one user
 with the openfire client(Spark) and yes it is connect to asterisk SIP also.
 But with the other one user, I register it with the SIP client(Zoiper/ or
 Linphone) and then I can make the call over these two SIP but they cannot
 reach the chat. I wonder what should I config between openfire and asterisk
 to enable chat over these two sip clients ?
 I am waiting for your reply, Thank.

 Thyda

 On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Good morning Thyda;

 Perhaps somebody has a solution for using it on Asterisk itself but
 after some trying I added the Openfire server as a IM server.

 I was a bit afraid that 'if' I got it working properly we had to
 maintain it and off course had to troubleshoot it in case it didn't work
 anymore.

 I've read something that you add a ams_msg context in extensions.conf
 but that didn't work for me unfortunaly. It did work for SIP Messages on
 phones but not for IM.

 I found Openfire easier to configure and it added a full integration
 with our LDAP which allowed single sign so that users could use the same
 password and log on automatically with the Jitsi client.

 But if you have some specific questions, I will be glad to answer.

 //Kristof
  Thyda ENG ength...@gmail.com 7/07/2015 6:07 
  I am currently, I create the VOIP server which enable the user to make
 the call over the asterisk server, Additionally now I want the user to be
 able to chat to each other too.
 I found some suggestion of using the openfire with asterisk but not much
 said on it, Anyway could you please share me how can I config the IM server
 over asterisk?

 I am waiting for your reply,

 Thyda

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Re: [asterisk-users] How to enable IM over the asterisk server

[James Cass]

The asterisk plugin for openfire would be what I would think would do that,
but as another person posted, it's very deprecated, so I'm not sure how
well it would work.  I've never used it personally.

James Cass http://goog_987864563
jcas...@gmail.com


On Wed, Jul 8, 2015 at 11:45 AM, Thyda ENG ength...@gmail.com wrote:

 Yes, I have though of setting them up on the same server(openfire, and
 asterisk) and the problem come in mind that how can register the user to
 openfire automatically when I register the user SIP on the asterisk server
 ? Do you have any idea? I am waiting for your reply.

 Thank,

 Thyda

 On Wed, Jul 8, 2015 at 6:55 PM, James Cass jcas...@gmail.com wrote:

 You can have the openfire server installed on the same server as asterisk
 without any issue, just size your server appropriately.  Just keep in mind
 they are different services.

 James Cass http://goog_987864563
 jcas...@gmail.com


 On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote:

 I just get started with it so my question maybe not well catch. Anyway
 to do the VOIP call and IM we need to use two difference servers? which one
 is asterisk for VOIP ? and other one for IM that is openfire ? or we can
 have other choice better than this ?
 Thank you for your help, I am waiting for your reply.

 Thyda


 On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Hi Thyda,

 I think you should see these as two individual systems. (I'm not an
 expert so just thinking out loud).

 Since you mention that you did a SIP mapping on Openfire, may I assume
 that you have the Asterisk IM plugin?

 In case of yes:
 Yes, there is a plugin between OpenFire and Asterisk but it is not
 actively developed anymore since 2006
 http://www.igniterealtime.org/projects/asterisk/

 So I don't think the plugin is really realiable anymore  on current
 versions.

 --

 I consider them as 2 separate systems which have to work on their own.
 Unfortunatly this means that every softphone has 2 accounts: one is SIP to
 Asterisk, one is XMPP to Openfire.

 That way our users are able to call internal/external using Asterisk,
 but do IM and internal calling via Openfire. (They can choose which source
 they take)

 Openfire is connected to our AD so our users just can logon with their
 Windows credentials.

 Unfortunatly, if you want a real production connection between Asterisk
 and Openfire, I'm unable to assist since I don't have the knowledge of it.
 sorry

 Hope this helps a bit.
 kristof
  Thyda ENG ength...@gmail.com 7/07/2015 11:28 
 Actually, I am using the openfire and I create two users with the SIP
 mapping on the openfire to the asterisk server. I can register one user
 with the openfire client(Spark) and yes it is connect to asterisk SIP also.
 But with the other one user, I register it with the SIP client(Zoiper/ or
 Linphone) and then I can make the call over these two SIP but they cannot
 reach the chat. I wonder what should I config between openfire and asterisk
 to enable chat over these two sip clients ?
 I am waiting for your reply, Thank.

 Thyda

 On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Good morning Thyda;

 Perhaps somebody has a solution for using it on Asterisk itself but
 after some trying I added the Openfire server as a IM server.

 I was a bit afraid that 'if' I got it working properly we had to
 maintain it and off course had to troubleshoot it in case it didn't work
 anymore.

 I've read something that you add a ams_msg context in extensions.conf
 but that didn't work for me unfortunaly. It did work for SIP Messages on
 phones but not for IM.

 I found Openfire easier to configure and it added a full integration
 with our LDAP which allowed single sign so that users could use the same
 password and log on automatically with the Jitsi client.

 But if you have some specific questions, I will be glad to answer.

 //Kristof
  Thyda ENG ength...@gmail.com 7/07/2015 6:07 
  I am currently, I create the VOIP server which enable the user to
 make the call over the asterisk server, Additionally now I want the user 
 to
 be able to chat to each other too.
 I found some suggestion of using the openfire with asterisk but not
 much said on it, Anyway could you please share me how can I config the IM
 server over asterisk?

 I am waiting for your reply,

 Thyda

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Re: [asterisk-users] small homebrew pbx

[James Cass]

I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php
for $30, and it works great for a home install.  Very low power draw as
well.

James Cass http://goog_987864563
jcas...@gmail.com


On Mon, Jun 15, 2015 at 10:50 AM, Kevin Larsen 
kevin.lar...@pioneerballoon.com wrote:

  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] sedwa...@sedwards.com causes me to be knocked off the list

[James Cass]

I agree with Doug, everything looks legit for sedwards.com from my point of
view.  Giles, maybe there's something wrong with the DNS server you're
using?

James Cass http://goog_987864563
jcas...@gmail.com


On Wed, Jun 3, 2015 at 11:01 AM, Giles Coochey gi...@coochey.net wrote:

 Someone on this list uses the address @sedwards.com

 I doubt this is their actual email address as there is no MX record for
 sedwards.com and I can't find registration for their domain either.

 Part of my mail servers reject these emails because they cannot be replied
 to, or are likely to be spam.

 Every so often I get a mail from the list management to say that I've been
 unsubscribed because of excessive bounces and it takes a single click to
 re-register.

 It's a bit of a niggle for me. What do you think I should do? Change my
 servers so that I don't check sender domains?



 --
 Regards,

 Giles Coochey, CCNP, CCNA, CCNAS
 NetSecSpec Ltd
 +44 (0) 8444 780677
 +44 (0) 7584 634135
 http://www.coochey.net
 http://www.netsecspec.co.uk
 gi...@coochey.net


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Re: [asterisk-users] Phones don't stop ringing when queue is answered

[James Thomas]

A few things I would try-
Change WaitExten to Wait(2)
Change Queue(queue_level_1,rtnC,18) to Queue(queue_level_1,rtnC,,,18)
Add an Answer() after the first Wait(2)
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Re: [asterisk-users] Phones don't stop ringing when queue is answered

[James Thomas]

What purpose do the WaitExten()s serve here? Are you really allowing the
caller to connect to different extensions in the test-queue context? Have
you tried without the WaitExten()s?
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Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[James Cass]

Akhilesh, for sip, just make sure you have up 5060 open and up 1-2
open in the security group to the public address where your phones will be
(soft or hard phones). You shouldn't need the address of your trunk
provider (since you should register to them). Of course you'll still need
22 open to manage the server, but you already have that. That's all you
need from a security group perspective.  Make sure that right then we can
inspect your config files.
On Apr 28, 2015 1:09 AM, akhilesh chand omakhileshch...@gmail.com wrote:

 Hi James,

 Please let me know how could I implement for sip.I will appreciate your
 help.

 Regards
 Akhilesh

 On Mon, Apr 27, 2015 at 7:01 PM, James Cass jcas...@gmail.com wrote:

 Akhilesh,
 I have implemented several ec2 instances with both sip and iax2 and have
 no problems with xlite or hard phones. Have you already opened the ports in
 the vpc security group on the Amazon side?  Let me know is I can help.
 --James
 On Apr 24, 2015 3:34 AM, akhilesh chand omakhileshch...@gmail.com
 wrote:

 Hi Thomas,

 Could you tell how can I change the protocol of corresponding port means
 5060 is configured with tcp protocol I want to configured with udp. When I
 execute nmap -p5060 xx.xx.xx.xx I got below output


 [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx

 Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC
 Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com
  (xx.xx.xx.xx)
 Host is up (0.00080s latency).
 PORT STATESERVICE
 5060/tcp filtered sip

 Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds

 Where I'm seeing 5060 is configured with tcp.


 Regards
 Akhilesh


 On Tue, Apr 21, 2015 at 5:10 PM, Thomas Stein himbe...@meine-oma.de
 wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Am 21.04.15 um 13:38 schrieb akhilesh chand:
  Hi Guenther,
 
  When  I executed nmap -p5060 xx.xx.xx.xx I got below output.
 
  [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx
 
  Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap
  scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com
  (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE
  SERVICE 5060/tcp filtered sip

 Maybe your softphone is trying UDP?

 cheers
 t.

  Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds
 
  On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter
  gboel...@gmail.com wrote:
 
  On 04/21/2015 04:58 PM, akhilesh chand wrote:
  Hi Guenther,
 
  What did you recommend to me, I did accordingly but there is no
  log showing in asterisk CLI. I'm getting same problem.
 
 
 
  Regards Akhilesh
 
  Hi Akhilesh,
 
  looks like your firewall is blocking it.
 
  Have you tried 'nmap -p5060 ip of your asterisk' or something
  similar?
 
  Regards Guenther
 
 
 
 
  On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter
  gboel...@gmail.com mailto:gboel...@gmail.com wrote:
 
  On 04/20/2015 12:31 PM, akhilesh chand wrote:
  Hi Folks,
 
  I'm trying to register softphone(X-lite) but I'm not able to
  register softphone whenever I'm trying to register softphone I
  got below error
 
  Inline image 1
 
  Is there any document/guide line where I will get process to
  register softphone in asterisk(Which is installed in EC2
  Cloud).
 
  Don't make it to complicated ...
 
  Connect to your Asterisk via ssh and run asterisk -rvv.
 
  Then let your Phone try to register. Asterisk should show you
  what's getting wrong.
 
  If you can't see anything while trying to register, shutdown
  your firewall and try it again ...
 
  Regards Guenther
 
 
 
  --
  _
 
 
 
  -- Bandwidth and Colocation Provided by http://www.api-digital.com
  --
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  Thurs: http://www.asterisk.org/hello
 
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  Thurs: http://www.asterisk.org/hello
 
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 --
 _
 -- Bandwidth and Colocation

Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).

[James Cass]

Akhilesh,
I have implemented several ec2 instances with both sip and iax2 and have no
problems with xlite or hard phones. Have you already opened the ports in
the vpc security group on the Amazon side?  Let me know is I can help.
--James
On Apr 24, 2015 3:34 AM, akhilesh chand omakhileshch...@gmail.com wrote:

 Hi Thomas,

 Could you tell how can I change the protocol of corresponding port means
 5060 is configured with tcp protocol I want to configured with udp. When I
 execute nmap -p5060 xx.xx.xx.xx I got below output


 [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx

 Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC
 Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com
  (xx.xx.xx.xx)
 Host is up (0.00080s latency).
 PORT STATESERVICE
 5060/tcp filtered sip

 Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds

 Where I'm seeing 5060 is configured with tcp.


 Regards
 Akhilesh


 On Tue, Apr 21, 2015 at 5:10 PM, Thomas Stein himbe...@meine-oma.de
 wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Am 21.04.15 um 13:38 schrieb akhilesh chand:
  Hi Guenther,
 
  When  I executed nmap -p5060 xx.xx.xx.xx I got below output.
 
  [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx
 
  Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap
  scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com
  (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE
  SERVICE 5060/tcp filtered sip

 Maybe your softphone is trying UDP?

 cheers
 t.

  Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds
 
  On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter
  gboel...@gmail.com wrote:
 
  On 04/21/2015 04:58 PM, akhilesh chand wrote:
  Hi Guenther,
 
  What did you recommend to me, I did accordingly but there is no
  log showing in asterisk CLI. I'm getting same problem.
 
 
 
  Regards Akhilesh
 
  Hi Akhilesh,
 
  looks like your firewall is blocking it.
 
  Have you tried 'nmap -p5060 ip of your asterisk' or something
  similar?
 
  Regards Guenther
 
 
 
 
  On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter
  gboel...@gmail.com mailto:gboel...@gmail.com wrote:
 
  On 04/20/2015 12:31 PM, akhilesh chand wrote:
  Hi Folks,
 
  I'm trying to register softphone(X-lite) but I'm not able to
  register softphone whenever I'm trying to register softphone I
  got below error
 
  Inline image 1
 
  Is there any document/guide line where I will get process to
  register softphone in asterisk(Which is installed in EC2
  Cloud).
 
  Don't make it to complicated ...
 
  Connect to your Asterisk via ssh and run asterisk -rvv.
 
  Then let your Phone try to register. Asterisk should show you
  what's getting wrong.
 
  If you can't see anything while trying to register, shutdown
  your firewall and try it again ...
 
  Regards Guenther
 
 
 
  --
  _
 
 
 
  -- Bandwidth and Colocation Provided by http://www.api-digital.com
  --
  New to Asterisk? Join us for a live introductory webinar every
  Thurs: http://www.asterisk.org/hello
 
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  visit: http://lists.digium.com/mailman/listinfo/asterisk-users
 
 
 
 
 
 
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 - -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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  Thurs: http://www.asterisk.org/hello
 
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 Version: GnuPG/MacGPG2 v2.0.16 (Darwin)

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 =tvB/
 -END PGP SIGNATURE-

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Re: [asterisk-users] Script to Program Snom Phones

[James B. Byrne]

On Thu, April 9, 2015 12:37, Tafadzwa Nyabasa wrote:
 Hi There,

 Does anyone know how to program Snom phones using a Mac addresses like
 what
 is done with the Ciscos. I have about 50 extensions to be programmed
 and I
 am hoping there is a way on Asterisk to program extensions on the snom
 phones. Please assist.

 Regards


I do not think that this is specifically an Asterisk problem.  The
SNOM phones that we use (870s and 76s) have FW 8.7.3.25.5.  On the
Update tab of the Advanced setting page there are set the update
policy and URI.  In our case the settings are 'Never update, load
settings only', from URL http://192.168.6.9:83, with a refresh
interval of 600840.

The phone will look at http://192.168.6.9:83 for a file called
snom870-.htm  where  is the phone's MAC
number.  If that fails then it will look for snom870.htm instead. 
These files should contain the xml dialect for the SNOM phone
configuration directives:

?xml version=1.0 encoding=utf-8?
settings
phone-settings
language perm=RWEnglish/language
dnd_on_code perm=*78/dnd_on_code
. . .
/phone-settings
/settings


You need to provide a service that will provide the file via URI. You
must put files therein with names following the specific nomenclature
employed buy the phones themselves. Finally you must also set the
phones to read from that location and to apply the configurations
retrieved therefrom.

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] Anonymous SIP calls

[James Cloos]

Some of us do allow sip from the internet, but just like for smtp email
protections are in order.

I point my SRV records at dedicated sip proxies (I use kamailio) which
check the INVITEd sip uri the same way my MXs check the SMTP Evelope-To
addresses, and only allow INVITEs through to authorized destinations.

And when those INVITEs make it to asterisk/freeswitch or the like, the
dialplan is generally not direct to phone(s), but via an IVR.

As an example, calling my email address via sip goes to an Asterisk
FollowMe instance.

I also provide my clients with dedicated sip addresses which avoid the
protections.

But the vast majority of the INVITEs coming to my public sip proxies are
fraud attempts.  My primary sip proxy has blocked over 32k fraudulent
INVITEs over the last six months.  And about one OPTIONS sip:100@... per
hour by something calling itself friendly-scanner.

Then again, the number of invalid sip INVITEs per public sip destination
are fewer than the number of spam/virus type SMTP attempts per unit time.

And all of the telemarking fraud I have had to deal with have come via
pstn dids, not via direct sip.

A half-gig virtual works fine for such a sip proxy.

You may also want to look into getting an ISN number, check out
http://freenum.org/ for the details.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] Anonymous SIP calls

[James B. Byrne]

On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
 You have to consider whether you really want anonymous calls, or you
 just want to enable SIP calls from trusted companies/partners.  The
 latter means setting up routes to these companies and (ideally)
 registration between peers.


This is what I am trying to get a handle on.  It seemed to me that the
promise of VOIP was essentially that one could use the Internet as a
replacement for the PSTN directly, providing that ones callers/callees
were also directly connected via VOIP.  SIP providers I had considered
a necessary transition to act as gateways between PSTN dialing and
VOIP until VOIP replaced PSTN virtually entirely if not completely.

That is why we are on Asterisk.  We had to replace our old keyed
system and the thought was that we might as well get ready for VOIP
even if we planned to stay on PSTN for the foreseeable future.

However, the overwhelming evidence I find is that one simply does not
employ VOIP in the same way that PSTN works.  Actually, I have put
that backwards.  What I have discovered is that the most commonly
recommended method is to switch from a Telco to A SIP provider and
continue in a manner similar to the former set-up.  External calls all
have to travel through a third party provider.

One does not accept incoming VOIP calls from just everyone,
apparently.  One only accepts VOIP calls from known correspondents.  I
am not clear why this is so other than vague warnings respecting
(admittedly real and serious) security issues.

Even limiting VOIP to known correspondents one is ultimately trusting
that they themselves are secured sufficiently to prevent unauthorised
access to your systems through theirs.  And that seems a bit of a
stretch by way of rationalisation to me.

Also I do not understand is why the same issues do not exist from
incoming calls via PSTN.

I somewhat understand the process of getting devices to register and
authenticate to obtain access to our outgoing routes.   What is it
about incoming SIP calls destined to our internal users that make
those calls so dangerous?  Why cannot incoming anonymous SIP calls not
be treated exactly as incoming PSTN calls (other than PSTN have to go
though DAHDI to turn them into digital VOIP calls). What is it that
prevents them from being blocked from gatewaying through to our PSTN
lines?

Please forgive my abysmal ignorance on this matter.  Perhaps I have
been down in the weeds too long getting our internal FreePBX system
working to see what is obvious to others.  I have been going theough
the Asticon Videos on security and have or already had implemented
most of the suggestions: Outbound LD secured by pins and allowed only
during work hours; IPTABLES rules and fail2ban checks; Separation of
voice and data network segments and addresses; Private IP for VOIP
desk-sets and internal provisioning; and so forth.

However, I still have the sense that I am just not getting it.  What
am I missing?

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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[asterisk-users] Anonymous SIP calls

[James B. Byrne]

We have a FreePBX-12 / Asterisk-12 setup that supports about 24
extensions, most internal Snom870s but six or so external (Jitsi-2.8).
 we use TLS and SRTP everywhere on our side of the fence.  The server
host is a dedicated atom(tm) box using the FreePBX distro (CentOS-6.x)
and is up-to-date.  Registrations require very long random passwords
and registrable devices are further restricted by netblock filters. 
We have the usual firewall and fail2ban intrusion prevention and
detection set-ups in place.

Our connection to the rest of the world is via PSTN.

We do our own DNS, both forward and reverse.  We have NAPTR and SRV
RRs for SIP and SIPS.

That is the environment.  Now for the questions.

Can I safely configure FreePBX/Asterisk to allow people to call us
directly via SIP?  In other words, sip://someth...@harte-lyne.ca would
reach us and ring internally as if someone had called our main office
number via PSTN.  Does it make sense to do so?

I am not talking about routing our main number through a SIP trunk
provider.  We will remain on PSTN for the foreseeable future.  But I
am curious as to whether or not it it worthwhile to allow others who
have the capability to simply call us via SIP rather than over PSTN. 
And if we do allow it what are the caveats and how does one actually
configure Asterisk to do it?

I have read a number of blogs, sections of the Definitive Asterisk
book and mailing list archived posts respecting anonymous SIP calls. 
But I have to say these leave me rather more confused than informed. 
Virtually all sources advise against accepting any anonymous incoming
SIP calls whatsoever.  The few that do not absolutely advise against 
do not give much guidance in how to handle incoming calls. And
frankly, I have only a dim idea how an incoming SIP call should be
handled from a theoretical point of view.

Any guidance would be welcome.


-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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[asterisk-users] Jitsi, SRTP and Asterisk 11

[James B. Byrne]

Does anyone here have a Jitsi softphone set up with Asterisk such that
SRTP is enabled, TLS is used to pass the SRTP key, and it works?
Anyone?  If so then what are the settings required for Asterisk?


-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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[asterisk-users] Guidence in DialPlan programming.

[James B. Byrne]

I am dealing with a FreePBX generated dialplan.  I have been following
the processing traces attempting to make use of the advice I received
here respecting setting a custom ring tone.   I have discovered that
the context I am using for incoming calls is not used at all during a
blind transfer.  Thus setting a third ring tone for that situation
inside that context is an impossibility.

I know now what I need to do and possibly where I need put it.  What I
wish is some guidance on how to properly return from my custom code
without damaging the dialplan elsewhere.

Here is the situation:

In extensions.conf I see this:
;-
;-

; Internal dialplan that most internal phones have access to
;
[from-internal]
include = from-internal-noxfer
include = from-internal-xfer
include = bad-number ; auto-generated
;-
;-
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible
; during a blind transfer, this context will not be visible
; during such.
;
[from-internal-noxfer]
include = from-internal-noxfer-custom
include = from-internal-noxfer-additional ; auto-generated
;-
;-
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during
; normal calls and blind transfers.
;
[from-internal-xfer]
include = from-internal-custom
include = from-internal-additional ; auto-generated
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
;-
;-




If a call is placed by a local extension then context
[from-internal-noxfer] is used.  If a blind transfer is performed the
context is [from-internal-xfer].  What I am considering is placing the
following code in extensions-custom.conf:



[from-internal-custom].
exten = _X,1,Noop()
exten = _X,n,Set(AlertSnom=http://www.notused.com\;info=)
exten = _X,n,Set(AlertInternalTransfer=alert_internal_transfer)
exten = _X,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})

exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)



My question is: are the last two lines the correct method of returning
from this back to extensions.conf?  Is there something else I should
use?  At them moment I just want to know how to properly and safely
return to the original referring context ([from-internal-xfer]).

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?

[James B. Byrne]

On Thu, March 5, 2015 09:56, Ruben Rögels wrote:


 Hi again,

 I'm glad to hear that I provided a somehow useful answer.

 Unfortunatelly, I don't know these details.
 If you wasn't lucky consulting the snom docs, maybe the snom support
 can be helpful with information about the exact implementation
 details.

 You also could use sip debug on asterisk to check what's going on
 when pressing the transfer button vs. what's happening when using
 ## via DTMF.

 Are you forced to get the transfer information from the SIP
 signalling, or can you use AMI events for example? I think
 this would be possible if asterisk is configured to stay in
 the media path, so re-inviting is handled over asterisk itself
 and therefore detectable with AMI events.


I am working with a FreePBX12/Asterisk11 setup.  Asterisk stays on the
path (B2B) and there are no peer-to-peer re-invites.

What I am trying to do is to get our Snom870s to use a distinctive
ring tone when external calls are transferred internally.  I have an
extension context override that detects the origin of calls and
assigns a distinctive ring to each based on ${CallerIDNum}.

But when a call is transferred then the tone does not change since the
CallerIDNum does not.  An external original call always rings as if it
were coming from the outside (which it is but transferred calls have a
different handling procedure than unanswered calls).  I need some way
to distinguish when the call has already been answered at least once
without changing the CallerID.

I am not worried about attended transfers since then the internal ring
tone is what should be used and that is what happens now.  I just need
to deal with blind transfers.

What I have now is:

1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring1 || ring2 (depending on 1 or 2)


What I want is:


1. Outside call = ring1
2. Internal call = ring2
3. Transferred call = ring3 (regardless of 1 or 2)


If everything went though ## then that would be simple enough.  The
trick is that most (all) users employ the transfer button and the
touch screen to forward calls using blind transfer.  But whatever
method they use to transfer I want the transfer ring tone to be the
same, albeit different from the one used for a new incoming call.

If the transfer is done using a sip message then that should be doable
as well.  I just have to discover what the message is.  If someone
already knows and would care to share the information then that would
be helpful.  Otherwise wireshark and debug will eventually reveal it.

I may not know what I am doing. But, at least I know that I do not
know what I am doing.

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?

[James B. Byrne]

On Thu, March 5, 2015 05:30, Ruben Rögels wrote:


 Am 05.03.2015 um 01:09 schrieb James B. Byrne:
 I am trying to determine how the transfer button on the Snom-870
 works
 with Asterisk.  Is the ## special code employed as when it is
 entered
 through the handset or is the blind transfer through the phone
 function accomplished in a different fashion?



 Hi,

 I hope I understood your question correctly.
 AFAIK, the transfer button sends a SIP message.
 Entering ## on the handset is recognized via DTMF by asterisk.


I hope that I understood what I was asking for.  Sometimes I do not.

  Yes, that is what I wanted to know.  Does the implementation of the
transfer button feature on the Snomp-870 use exactly the same
technique as the ## feature code entered through the dial pad and
produce exactly the same SIP message that Asterisk produces when it
gets the ## DTMF?

The reason is that I wish to be able to detect a call transfer
performed via either method (## or Transfer-Button) and process the
result of both in the same fashion. If the button and DTMF transfers
are not performed using the same switching techniques in Asterisk then
I need to discover what those differences are.  If both are totally
equivalent from a SIP message signalling point of view then the issue
is far easier to handle.

I searched, in vain, in the Snom-870 docs for specifics on this and
either could not find or did not recognize anything that applied.  Do
you know where I can locate these sorts of details.  My knowledge of
SIP/RTP/VOIP etc. is cursory but, given an adequate reference, I can
usually sort things out.

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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[asterisk-users] OT - How does the blind transfer function work on Snom-870?

[James B. Byrne]

I am trying to determine how the transfer button on the Snom-870 works
with Asterisk.  Is the ## special code employed as when it is entered
through the handset or is the blind transfer through the phone
function accomplished in a different fashion?


-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

This seems to me to be getting down to some sort of problem with
configuring the Snom-870.

when I register the device 41712 (set up for transport=tls only) then
I see this in the SIP trace:


Sent to udp:192.168.6.9:5060 at 4/3/2015 09:07:36:813 (836 bytes):

REGISTER sip:voinet09.internal.hamilton.harte-lyne.ca:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.112:5060;branch=z9hG4bK-udx92poqese6;rport
From: James B Byrne
sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061;tag=frgaimnglp
To: James B Byrne
sip:41...@voinet09.internal.hamilton.harte-lyne.ca:5061
Call-ID: 71004941-gk6y4evf6dci
CSeq: 482 REGISTER
Max-Forwards: 70
Contact:
sip:41712@192.168.6.112:5060;line=0p8zx4sh;reg-id=1;q=1.0;+sip.instance=urn:uuid:ad1349a7-e08d-411b-83b0-000413281B56;audio;mobility=fixed;duplex=full;description=snom870;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO
User-Agent: snom870/8.7.3.25.5
Allow-Events: dialog
X-Real-IP: 192.168.6.112
Supported: path, gruu
Expires: 3600
Content-Length: 0


The SNOM-870 is sending registration via UDP and not by TLS.  Is that
how things are supposed to work?  If only TLS is enabled in Asterisk
for that peer then evidently the peer cannot register.  But is
registration supposed to be done via TLS?  If so then how does one
configure the Snom to do so?

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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[asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

CentOS-6.5 (FreePBX-2.6)
Asterisk-11.14.2 (FreePBX)
snom870-SIP 8.7.3.25.5

I am having a very difficult time attempting to get TLS and SRTP
working with Asterisk and anything else.  At the moment I am trying to
get TLS functioning with our Snom870 desk-sets.  And I am not having
much luck.

Since this is an extraordinarily (to me) Byzantine environemnt I am
going to ask if any of you have gotten this set-up (Asterisk11 with
Snom870s using TLS) to work and if so could you provide the details?

I have this in Asterisk sip.conf (loaded through FreePBXs
sip_general_additional.conf).

tcpenable=yes
tlsenable=yes
tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
tlsdontverifyserver=yes
tlscipher=ALL
tlsclientmethod=tlsv1

And I have this for the test device context:

[41712]
deny=0.0.0.0/0.0.0.0
secret=NearlyANastyThat
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=tls,udp,tcp
avpf=no
force_avp=no
icesupport=no
encryption=yes
callgroup=
pickupgroup=
dial=SIP/41712
mailbox=41712@device
permit=192.168.6.0/255.255.255.0
callerid=James B Byrne 41712
callcounter=yes
faxdetect=no
cc_monitor_policy=generic

If I change the transport setting to TLS then I get this reported:

[2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875
ast_tcptls_client_start: Unable to connect SIP socket to
192.168.6.112:5060: Connection refused

I cannot seem to configure the Snom870 to listen for TCP on 5060. 
There is a setting for that on the phone but it seems to have no
effect (it always returns to NO following a reboot). The Snom website
says that the option is not available in FW8.5 and later. It does not
inform one of whether that the phone listens by default or not on
FW8.5+, only that the option has no effect.

It also does not say, as far as I can find, whether Snom870s listen
for TCP at all or on what port.  One may infer that since these
devices purport to support TLS that the answer is yes and that TCP5061
is a likely candidate.  But they do not seem to come right out and say
so anywhere.

In a section devoted to the Snom370, which is a model that we do not
employ, there is reference to DNS SRV RRs.  The inference drawn from
the examples given is that these will control what ports the Snom will
listen on for which services.

We have such records in our DNS zone. They look like this:

;# Configure sip/sips service records (VOIP)
;HOST   TTL CLASS   TYPEORDER   PREF
FLAGS   SERVICE REGEXP  REPLACEMENT

300 IN  NAPTR   50  50  
s SIPS+D2T_sips._tcp.harte-lyne.ca.

300 IN  NAPTR   90  50  
s SIP+D2T _sip._tcp.harte-lyne.ca.

300 IN  NAPTR   100 50  
s SIP+D2U _sip._udp.harte-lyne.ca.

;HOST   TTL CLASS   TYPEORDER   PREF
PORTTARGET

_sips._tcp.harte-lyne.ca.   300 IN  SRV 10  10  
5061voinet09.hamilton.harte-lyne.ca.

_sip._tcp.harte-lyne.ca.300 IN  SRV 10  10  
5060voinet09.hamilton.harte-lyne.ca.

_sip._udp.harte-lyne.ca.300 IN  SRV 10  10  
5060voinet09.hamilton.harte-lyne.ca.

However, our phones are configured to use SIP accounts having the form
account@ipv4-addr.  I doubt greatly that the Snom870s will perform a
reverse DNS lookup on the provider's IPv4 to discover the forward zone
domain and thus I do not believe that SRV RRs can help us in this
instance.  They certainly do not seem to have any effect.

Asterisk seems not to distinguish between 5060 and 5061 regarless of
protocol.  I am not sure then how to proceed.  Is there a way to force
Asterisk to talk to port TCP5061 on a specific device?  Is this an
exclusive setting?

This long background is by way of asking for help.  If I have not
provided specific information that is significant to this problem then
I will do so if asked.

What I am attempting has to be possible.  Somehow.  And somebody must
have already accomplished this. Somewhere.

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James Cloos]

 JBB == James B Byrne byrn...@harte-lyne.ca writes:

JBB tcpenable=yes
JBB tlsenable=yes
JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt
JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt
JBB tlsdontverifyserver=yes
JBB tlscipher=ALL
JBB tlsclientmethod=tlsv1

You are missing the tls key.

The config name is tlsprivatekey; set that to the filename of your tls
key, akin to how tlscertfile is set.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

These are the sip settings on our installion.

Global Settings:

  UDP Bindaddress:0.0.0.0:5060
  TCP SIP Bindaddress:0.0.0.0:5060
  TLS SIP Bindaddress:(null)
  Videosupport:   No
  Textsupport:No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:Off
  Match Auth Username:No
  Allow unknown access:   Yes
  Allow subscriptions:Yes
  Allow overlap dialing:  Yes
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support: No
  Realm. auth:No
  Our auth realm  asterisk
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:Yes
  Direct RTP setup:   No
  User Agent: FPBX-12.0.40(11.14.2)
  SDP Session Name:   Asterisk PBX 11.14.2
  SDP Owner Name: root
  Reg. context:   (not set)
  Regexten on Qualify:No
  Trust RPID: No
  Send RPID:  No
  Legacy userfield parse: No
  Send Diversion: Yes
  Caller ID:  Unknown
  From: Domain:
  Record SIP history: Off
  Call Events:On
  Auth. Failure Events:   Off
  T.38 support:   No
  T.38 EC mode:   Unknown
  T.38 MaxDtgrm:  4294967295
  SIP realtime:   Disabled
  Qualify Freq :  6 ms
  Q.850 Reason header:No
  Store SIP_CAUSE:No

Network QoS Settings:
---
  IP ToS SIP: CS3
  IP ToS RTP audio:   EF
  IP ToS RTP video:   AF41
  IP ToS RTP text:CS0
  802.1p CoS SIP: 4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:5
  Jitterbuffer enabled:   No

Network Settings:
---
  SIP address remapping:  Enabled using externaddr
  Externhost: none
  Externaddr: 216.185.71.9:0
  Externrefresh:  10
  Localnet:   216.185.71.0/255.255.255.0
  192.168.6.0/255.255.255.0
  192.168.209.0/255.255.255.0
  192.168.216.0/255.255.255.0
  192.168.71.0/255.255.255.0

Global Signalling Settings:
---
  Codecs: (gsm|ulaw|alaw)
  Codec Order:ulaw:20,alaw:20,gsm:20
  Relax DTMF: No
  RFC2833 Compensation:   No
  Symmetric RTP:  Yes
  Compact SIP headers:No
  RTP Keepalive:  0 (Disabled)
  RTP Timeout:30
  RTP Hold Timeout:   300
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup: No
  Pedantic SIP support:   Yes
  Reg. min duration   60 secs
  Reg. max duration:  3600 secs
  Reg. default duration:  120 secs
  Sub. min duration   60 secs
  Sub. max duration:  3600 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Outbound reg. retry 403:0
  Notify ringing state:   Yes
Include CID:  No
  Notify hold state:  Yes
  SIP Transfer mode:  open
  Max Call Bitrate:   384 kbps
  Auto-Framing:   No
  Outb. proxy:not set
  Session Timers: Accept
  Session Refresher:  uas
  Session Expires:1800 secs
  Session Min-SE: 90 secs
  Timer T1:   500
  Timer T1 minimum:   100
  Timer B:32000
  No premature media: Yes
  Max forwards:   70

Default Settings:
-
  Allowed transports: UDP
  Outbound transport: UDP
  Context:from-sip-external
  Record on feature:  automon
  Record off feature: automon
  Force rport:Yes
  DTMF:   rfc2833
  Qualify:0
  Keepalive:  0
  Use ClientCode: No
  Progress inband:Never
  Language:
  Tone zone:  Not set
  MOH Interpret:  default
  MOH Suggest:
  Voice Mail Extension:   *97

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

On Tue, March 3, 2015 13:19, jg wrote:

 Forget about the reverse DNS stuff for the moment.

 Do simple SIP accounts (without SRTP/SRTP and deny/permit stuff) work?

 Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag.

 Then try the rest.

 jg


The Snom870s and our Asterisk FreePBX are communicating with each
other and have been for the past two years.  The Snoms are configured
for AES-80 and SRTP is enabled on the FreePBX device entry. We have a
working PBX system.  I am trying to secure it.

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James Cloos]

Other things to consider:

The transport config, which can be in [general] or in a peer's [] block.
if you want tls-only, use transport=tls
it also accepts tcp, udp or a comma-separated list.
if given a list, it tries them in order

If you need ast to register over tls, use something like this:

   register = tls://username:xxx...@sip-tls-proxy.example.org

(copied from the example sip.conf).

Set tlsbindaddr to the address to which to bind(2) the tls socket.
tlsbindaddr=0.0.0.0 is typical in ipv4-only configs.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

I reconfigured sip.conf to have these settings:

tcpenable=yes
tlsenable=yes
tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
tlsdontverifyserver=yes
tlscipher=ALL
tlsclientmethod=tlsv1
tlsprivatekey=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.key
tcpbindaddr=0.0.0.0/0.0.0.0:5061
tlsbindaddr=0.0.0.0/0.0.0.0:5061

Following amportal a r I see this:


[2015-03-03 16:26:48] ERROR[17130]: tcptls.c:875
ast_tcptls_client_start: Unable to connect SIP socket to
192.168.6.112:5060: Connection refused

This is what sip show settings reveals:


Global Settings:

  UDP Bindaddress:0.0.0.0:5060
  TCP SIP Bindaddress:0.0.0.0:5060
  TLS SIP Bindaddress:0.0.0.0:5061


Is it just me or is there something odd about specifying a TCP port
and then having it ignored?



-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

On Tue, March 3, 2015 16:34, James Cloos wrote:
 Other things to consider:

 The transport config, which can be in [general] or in a peer's []
 block.
 if you want tls-only, use transport=tls
 it also accepts tcp, udp or a comma-separated list.
 if given a list, it tries them in order


The specific device I am using to test this with has only
transport=tls set.  Which is why it cannot register because the
default fall-back to udp is not permitted.

 If you need ast to register over tls, use something like this:

register = tls://username:xxx...@sip-tls-proxy.example.org

Does this go in the device context?  In other words is it placed in
the same context that the device's transport value is set?  Would the
following be valid?

[device]
register = tls://user:extension@192.168.6.112:5061


How would multiple users at a single device be handled?


 (copied from the example sip.conf).

 Set tlsbindaddr to the address to which to bind(2) the tls socket.
 tlsbindaddr=0.0.0.0 is typical in ipv4-only configs.

 -JimC

Presumably this is equivalent to tlsbindaddr=0.0.0.0/0.0.0.0?  Is the
syntax tlsbindaddr=0.0.0.0/0.0.0.0:5061 is also correct?


-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] TLS, SRTP, Asterisk11 and Snom870s

[James B. Byrne]

On Tue, March 3, 2015 13:37, James Cloos wrote:
 JBB == James B Byrne byrn...@harte-lyne.ca writes:

 JBB tcpenable=yes
 JBB tlsenable=yes
 JBB tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt
 JBB tlscafile=/etc/pki/tls/certs/ca-bundle.crt
 JBB tlsdontverifyserver=yes
 JBB tlscipher=ALL
 JBB tlsclientmethod=tlsv1

 You are missing the tls key.

 The config name is tlsprivatekey; set that to the filename of your tls
 key, akin to how tlscertfile is set.

 -JimC

Thank you.  The settings in sip_general_additional.conf are now:

tcpenable=yes
tlsenable=yes
tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.pem
tlscafile=/etc/pki/tls/certs/ca-bundle.crt
tlsdontverifyserver=yes
tlscipher=ALL
tlsclientmethod=tlsv1
tlsprivatekey=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.key


However, issuing 'amportal a r' still results in this error:



[2015-03-03 15:40:42] ERROR[13681]: tcptls.c:875
ast_tcptls_client_start: Unable to connect SIP socket to
192.168.6.112:5060: Connection refused

-- 
***  E-Mail is NOT a SECURE channel  ***
James B. Byrnemailto:byrn...@harte-lyne.ca
Harte  Lyne Limited  http://www.harte-lyne.ca
9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
Canada  L8E 3C3


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Re: [asterisk-users] Weird SIP stuff

[James Cloos]

 EW == Eric Wieling ewiel...@nyigc.com writes:

EW Basically, the traffic is coming from 65.211.180.237 but the header is:

EW f: 
sip:+1347545xxx@199.173.94.80:5060;user=phone;tag=4-45026-159e4a6-995949f-159e4a6

The From: header and the socket ip are often different.

Check for Via: headers.  Those show the path of the SIP.  In your
example, look for a Via which mentions 65.211.180.237.

Note that the From does not necessary have to be a sip address reachable
from the outside.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 0x997A9F17ED7DAEA6

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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

On Wed, Nov 26, 2014 at 3:20 PM, James Lamanna jlama...@gmail.com wrote:


 On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com
 wrote:


 On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com
 wrote:

 On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com
 wrote:
  Also, how big does the cache in frame.c grow to?
  I've recompiled with MALLOC_DEBUG on that server:
 
  asterisk -rx memory show summary
 
  
  1780466242 bytes (1780181594 cache) in2352909 allocations in file
  frame.c
  ...
 
  Seems like a ridiculous cache.
 

 I'm not going to respond to your new thread, since it is the same
 discussion as this one.

 The frame cache is a per-thread local cache of frames that prevents
 having to re-allocate frames as they pass through Asterisk. Clearly,
 something is abusing it.

 I think you'll need to provide some more information on how you're
 producing this situation. Specifically:
  * Channel technologies involved, and the formats on the channels
  * Dialplan that reproduces the problem

 Are you using any non-core dialplan applications or channel drivers?


 This PBX has about 100 registered SIP clients, along with 23 PRI
 channels, 2 inbound/outbound SIP trunks and around 100 IAXModems registered
 to it. It primarily handles faxing.
 I am not using any non-standard channel drivers. I am using the T.38
 gateway funcionality.

 The jist of the dialplan is this: (example of the PRI and a SIP trunk,
 inbound)

 [pri-in]
 exten = _X.,1,Set(__FROM_DID=${EXTEN})
 exten = _X.,n,Set(FAX_IDX=700)
 exten = _X.,n,Set(MAX_IDX=719)
 exten = _X.,n,Goto(dial-hylafax,s,1)

 [sip-trunk-in]
 exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
 exten = _X.,n,Set(FAX_IDX=950)
 exten = _X.,n,Set(MAX_IDX=959)
 exten = _X.,n,Set(FAXOPT(gateway)=yes)
 exten = _X.,n,Goto(dial-hylafax,s,1)

 [dial-hylafax]
 exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
 exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
 exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
 exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
 exten = s,n,NoOp(${STATE})
 exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
 exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
 exten = s,n,Goto(s-${DIALSTATUS},1)
 exten = s,n(nofax),Playtones(busy)
 exten = s,n,NoOp(NO MODEMS AVAILABLE)
 exten = s,n,Wait(20)
 exten = s,n,Hangup()
 exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
 exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
 exten = s-ANSWER,n,Hangup()
 exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
 exten = _s-.,n,Goto(s,1)
 exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)

 The current state requires me to restart Asterisk almost every day.
 I'm also seeing this on a completely different machine after upgrading
 from Asterisk10 to 11.


 I'm wondering if this is a problem in the SLIN converter?
 I do use SLIN with iaxmodem.


Also of note,
A quick valgrind run and attempting to send a few faxes produces a bunch of
these in the valgrind output:

==30640== 217,259 bytes in 287 blocks are definitely lost in loss record
1,778 of 1,789
==30640==at 0x4C267CC: calloc (vg_replace_malloc.c:467)
==30640==by 0x4DC50E: ast_frdup (utils.h:523)
==30640==by 0x47125F: __ast_queue_frame (channel.c:1284)
==30640==by 0x1EF75589: __do_deliver (chan_iax2.c:3102)
==30640==by 0x1EF76C5A: schedule_delivery (chan_iax2.c:4374)
==30640==by 0x1EF8F497: socket_process_helper (chan_iax2.c:12010)
==30640==by 0x1EF99C37: iax2_process_thread (chan_iax2.c:12030)
==30640==by 0x56C458: dummy_start (utils.c:1192)
==30640==by 0x5E359C9: start_thread (pthread_create.c:300)
==30640==by 0x270326FF: ???
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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

On Tue, Nov 25, 2014 at 10:21 AM, James Lamanna jlama...@gmail.com wrote:


 On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com
 wrote:

 On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com
 wrote:
  Also, how big does the cache in frame.c grow to?
  I've recompiled with MALLOC_DEBUG on that server:
 
  asterisk -rx memory show summary
 
  
  1780466242 bytes (1780181594 cache) in2352909 allocations in file
  frame.c
  ...
 
  Seems like a ridiculous cache.
 

 I'm not going to respond to your new thread, since it is the same
 discussion as this one.

 The frame cache is a per-thread local cache of frames that prevents
 having to re-allocate frames as they pass through Asterisk. Clearly,
 something is abusing it.

 I think you'll need to provide some more information on how you're
 producing this situation. Specifically:
  * Channel technologies involved, and the formats on the channels
  * Dialplan that reproduces the problem

 Are you using any non-core dialplan applications or channel drivers?


 This PBX has about 100 registered SIP clients, along with 23 PRI channels,
 2 inbound/outbound SIP trunks and around 100 IAXModems registered to it. It
 primarily handles faxing.
 I am not using any non-standard channel drivers. I am using the T.38
 gateway funcionality.

 The jist of the dialplan is this: (example of the PRI and a SIP trunk,
 inbound)

 [pri-in]
 exten = _X.,1,Set(__FROM_DID=${EXTEN})
 exten = _X.,n,Set(FAX_IDX=700)
 exten = _X.,n,Set(MAX_IDX=719)
 exten = _X.,n,Goto(dial-hylafax,s,1)

 [sip-trunk-in]
 exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
 exten = _X.,n,Set(FAX_IDX=950)
 exten = _X.,n,Set(MAX_IDX=959)
 exten = _X.,n,Set(FAXOPT(gateway)=yes)
 exten = _X.,n,Goto(dial-hylafax,s,1)

 [dial-hylafax]
 exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
 exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
 exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
 exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
 exten = s,n,NoOp(${STATE})
 exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
 exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
 exten = s,n,Goto(s-${DIALSTATUS},1)
 exten = s,n(nofax),Playtones(busy)
 exten = s,n,NoOp(NO MODEMS AVAILABLE)
 exten = s,n,Wait(20)
 exten = s,n,Hangup()
 exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
 exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
 exten = s-ANSWER,n,Hangup()
 exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
 exten = _s-.,n,Goto(s,1)
 exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)

 The current state requires me to restart Asterisk almost every day.
 I'm also seeing this on a completely different machine after upgrading
 from Asterisk10 to 11.


I'm wondering if this is a problem in the SLIN converter?
I do use SLIN with iaxmodem.

-- James
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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan mjor...@digium.com wrote:

 On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna jlama...@gmail.com wrote:
  Also, how big does the cache in frame.c grow to?
  I've recompiled with MALLOC_DEBUG on that server:
 
  asterisk -rx memory show summary
 
  
  1780466242 bytes (1780181594 cache) in2352909 allocations in file
  frame.c
  ...
 
  Seems like a ridiculous cache.
 

 I'm not going to respond to your new thread, since it is the same
 discussion as this one.

 The frame cache is a per-thread local cache of frames that prevents
 having to re-allocate frames as they pass through Asterisk. Clearly,
 something is abusing it.

 I think you'll need to provide some more information on how you're
 producing this situation. Specifically:
  * Channel technologies involved, and the formats on the channels
  * Dialplan that reproduces the problem

 Are you using any non-core dialplan applications or channel drivers?


This PBX has about 100 registered SIP clients, along with 23 PRI channels,
2 inbound/outbound SIP trunks and around 100 IAXModems registered to it. It
primarily handles faxing.
I am not using any non-standard channel drivers. I am using the T.38
gateway funcionality.

The jist of the dialplan is this: (example of the PRI and a SIP trunk,
inbound)

[pri-in]
exten = _X.,1,Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=700)
exten = _X.,n,Set(MAX_IDX=719)
exten = _X.,n,Goto(dial-hylafax,s,1)

[sip-trunk-in]
exten = _X.,1(normal),Set(__FROM_DID=${EXTEN})
exten = _X.,n,Set(FAX_IDX=950)
exten = _X.,n,Set(MAX_IDX=959)
exten = _X.,n,Set(FAXOPT(gateway)=yes)
exten = _X.,n,Goto(dial-hylafax,s,1)

[dial-hylafax]
exten = s,1,GotoIf($[${FROM_DID:0:1} = 1]?prune:cont)
exten = s,n(prune),Set(__FROM_DID=${FROM_DID:1})
exten = s,n(cont),GotoIf($[${FAX_IDX} = ${MAX_IDX}]?tryfax:nofax)
exten = s,n(tryfax),Set(STATE=${DEVICE_STATE(Custom:iaxmodem${FAX_IDX})})
exten = s,n,NoOp(${STATE})
exten = s,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=INUSE)
exten = s,n,Dial(IAX2/iaxmodem${FAX_IDX}/${FROM_DID},60,g)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s,n(nofax),Playtones(busy)
exten = s,n,NoOp(NO MODEMS AVAILABLE)
exten = s,n,Wait(20)
exten = s,n,Hangup()
exten = s-ANSWER,1,NoOp(IAXMODEM HANGUP)
exten = s-ANSWER,n,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)
exten = s-ANSWER,n,Hangup()
exten = _s-.,1,Set(FAX_IDX=${MATH(1+${FAX_IDX},i)})
exten = _s-.,n,Goto(s,1)
exten = h,1,Set(DEVICE_STATE(Custom:iaxmodem${FAX_IDX})=NOT_INUSE)

The current state requires me to restart Asterisk almost every day.
I'm also seeing this on a completely different machine after upgrading from
Asterisk10 to 11.

-- James
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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

cat /proc/cpuinfo lists 4 cores.
So even if that's not showing hyperthreading, maximum 8.
By your rule, that would be 8 cores * 0.5GB = 4GB memory.
I've seen resident memory be up over 6GB.

On Sat, Nov 22, 2014 at 1:29 PM, Freddi Hansen f...@danovation.dk wrote:


 Its up to 5.8G of resident memory with 28321 calls processed.
 The OOM killer is going to kill this soon at this rate (8GB RAM machine).
 This seems like a pretty serious problem.
 It looks like I'll need to restart asterisk every night

 Hi the number of cpu cores that you see with top  times 512Mbyte is the
 level of ram that's needed

 e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be (
 2 x 8 x 2  x 0,5 gb ) = 16 gb  + a bit exstra.
 So from start memory usage increases until it reaches 17.3 gb and then
 stabilizes. at that level.
 You can disables hypertreading and cut your ram usage to half of that.

 I can't see what hardware you are using but I think you need to check that
 the rule above fits your hardware.

 b.r.
 Freddi






 On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com
 wrote:
 Hi,
 I have an Asterisk server that's been running now for around 2 days.
 I've noticed that the resident memory seems to be very high for its
 current call load:

PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
   18321 asterisk  20   0 8050m 5.2g 6968 S   13
 66.2 363:11.80 asterisk

 $ asterisk -rx core show channels

 24 active channels

 12 active calls

 25216 calls processed


  This server has a bunch of IAXModems hooked up to it and is mainly used
 as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2
 days with only 12 currently active calls?

 I am not using any realtime peers.

 There are 100 registered SIP peers on this server as well.

 Thanks.

 -- James


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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

Also, how big does the cache in frame.c grow to?
I've recompiled with MALLOC_DEBUG on that server:

asterisk -rx memory show summary


1780466242 bytes (1780181594 cache) in2352909 allocations in file
frame.c
...

Seems like a ridiculous cache.



On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna jlama...@gmail.com wrote:

 cat /proc/cpuinfo lists 4 cores.
 So even if that's not showing hyperthreading, maximum 8.
 By your rule, that would be 8 cores * 0.5GB = 4GB memory.
 I've seen resident memory be up over 6GB.

 On Sat, Nov 22, 2014 at 1:29 PM, Freddi Hansen f...@danovation.dk wrote:


 Its up to 5.8G of resident memory with 28321 calls processed.
 The OOM killer is going to kill this soon at this rate (8GB RAM machine).
 This seems like a pretty serious problem.
 It looks like I'll need to restart asterisk every night

 Hi the number of cpu cores that you see with top  times 512Mbyte is the
 level of ram that's needed

 e.g. a hp-gen8 with 2 octo core cpu's and hyperthreading enabled will be
 ( 2 x 8 x 2  x 0,5 gb ) = 16 gb  + a bit exstra.
 So from start memory usage increases until it reaches 17.3 gb and then
 stabilizes. at that level.
 You can disables hypertreading and cut your ram usage to half of that.

 I can't see what hardware you are using but I think you need to check
 that the rule above fits your hardware.

 b.r.
 Freddi






 On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com
 wrote:
 Hi,
 I have an Asterisk server that's been running now for around 2 days.
 I've noticed that the resident memory seems to be very high for its
 current call load:

PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 18321 asterisk  20   0 8050m 5.2g 6968 S   13
 66.2 363:11.80 asterisk

 $ asterisk -rx core show channels

 24 active channels

 12 active calls

 25216 calls processed


  This server has a bunch of IAXModems hooked up to it and is mainly used
 as a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2
 days with only 12 currently active calls?

 I am not using any realtime peers.

 There are 100 registered SIP peers on this server as well.

 Thanks.

 -- James


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[asterisk-users] Size of frame.c cache in Asterisk 11?

[James Lamanna]

(Starting a new email topic for this specific issue)

Hi,
What is the maximum size of the frame.c cache in Asterisk 11 and why does
it constantly increase?

This is what I'm up to already:

$ asterisk -rx memory show summary
3667584471 bytes (3667366799 cache) in4846685 allocations in file
frame.c

~$ asterisk -rx core show uptime
System uptime: 2 days, 11 hours, 12 minutes, 12 seconds
Last reload: 2 days, 11 hours, 12 minutes, 12 seconds

$ asterisk -rx core show channels
34 active channels
17 active calls
13824 calls processed

This seems like very odd behavior.

Thanks.

-- James
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[asterisk-users] Dahdi fxo vs sip blf

[James Cloos]

It has been may years since I've done anything with a dahdi fxo; much
has changed in the interim and I havne't found answers googling.

The fxo hw is installed on the pots line in parallel to existing pots
phones.

My goal is to have a blf on the sip phone which lights whenever any of
the devices on the pots line are off hook and which, when pressed,
INVITEs the asterisk box such that it takes the fxo off hook and joins
the existing conversation.  Or just passes the dialtone over the rtp if
eerything was on hook when the blf was hit.

Clearly Hint is part of the solution, but my attempts so far have
failed.

At the moment I'm on 12.7, but still using chan_sip.  Converting the
chan_pjsip will be the next project for this box.

What is the proper way to set this up?

-JimC
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[asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

Hi,
I have an Asterisk server that's been running now for around 2 days.
I've noticed that the resident memory seems to be very high for its current
call load:

  PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
18321 asterisk  20   0 8050m 5.2g 6968 S   13 66.2
363:11.80 asterisk

$ asterisk -rx core show channels

24 active channels

12 active calls

25216 calls processed


This server has a bunch of IAXModems hooked up to it and is mainly used as
a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2
days with only 12 currently active calls?

I am not using any realtime peers.

There are 100 registered SIP peers on this server as well.

Thanks.

-- James
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Re: [asterisk-users] High resident memory with 11.14.0 ?

[James Lamanna]

Its up to 5.8G of resident memory with 28321 calls processed.
The OOM killer is going to kill this soon at this rate (8GB RAM machine).
This seems like a pretty serious problem.
It looks like I'll need to restart asterisk every night



On Fri, Nov 21, 2014 at 10:53 AM, James Lamanna jlama...@gmail.com wrote:

 Hi,
 I have an Asterisk server that's been running now for around 2 days.
 I've noticed that the resident memory seems to be very high for its
 current call load:

   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
 18321 asterisk  20   0 8050m 5.2g 6968 S   13 66.2
 363:11.80 asterisk

 $ asterisk -rx core show channels

 24 active channels

 12 active calls

 25216 calls processed


 This server has a bunch of IAXModems hooked up to it and is mainly used as
 a Fax gateway to hylafax. Is this normal? 5.2Gig of memory used after 2
 days with only 12 currently active calls?

 I am not using any realtime peers.

 There are 100 registered SIP peers on this server as well.

 Thanks.

 -- James



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Re: [asterisk-users] Setting channel musicclass from AGI

[James Lamanna]

Hi Matt,
So this actually works (haven't had a chance to try it)?

SET VARIABLE CHANNEL(musicclass) default

Because musicclass is piece of channel information.
Referencing ${musicclass} is not the same thing.

Thanks.

-- James

On Sun, Oct 5, 2014 at 8:05 PM, Matthew Jordan mjor...@digium.com wrote:

 On Sun, Oct 5, 2014 at 6:40 PM, James Lamanna jlama...@gmail.com wrote:
  Hi,
  Since SetMusicOnHold() is being deprecated, how do we set the channel
  musicclass from an AGI script?
  Last time I checked you can't call dialplan functions from AGI.
 

 Actually, you can. Any time you can evaluate or set a channel
 variable, you can also evaluate or set a dialplan function. Hence, you
 can use both 'get variable' [1] or 'set variable' [2]. You could also
 use 'exec' and call the Set dialplan application directly.

 [1] https://wiki.asterisk.org/wiki/display/AST/AGICommand_get+variable
 [2] https://wiki.asterisk.org/wiki/display/AST/AGICommand_set+variable

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 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org

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[asterisk-users] Setting channel musicclass from AGI

[James Lamanna]

Hi,
Since SetMusicOnHold() is being deprecated, how do we set the channel
musicclass from an AGI script?
Last time I checked you can't call dialplan functions from AGI.

Thanks.

-- James
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Re: [asterisk-users] Show Log(NOTICE) messages on the console

[James Thomas]

logger.conf... You should start by comparing that file between the two
servers. Not sure if it's still called logger.conf in asterisk 11 though.
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Re: [asterisk-users] if statement recording - after hours

[James Thomas]

GotoIfTime()
Check out-
http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime

If the time is within a certain range, execute the recording dialplan. If
it's outside the range, then skip to the dialplan after the recording stuff.
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Re: [asterisk-users] Asterisk-eSpeak and Asterisk 12

[James Cloos]

 O == Olivier  oza.4...@gmail.com writes:

O Hijacking this thread, [the espeak] default voice ... produces a
O quite rebotic voice.  Is this something that can be improved...?

O exten = 1234,n,Espeak(This is a simple espeak test in english.,any)

Try the other voices.

Run espeak from the command line, try out the various voices and the
other options which also are available in the Asterisk-eSpeak espeak.conf
file until you find a combination which sounds OK.

Depending on your hardware and settings, you may need to have espeak
write its output to a file, and use another applicaion to play that.

The option space for espeak has large variability.

Flite also needs such tuning for nice output.

-JimC
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Re: [asterisk-users] SIP trunk gives fuzzy / distorted audio on mobiles, OK on fixed lines

[James Thomas]

Is the quality the same incoming from mobile as outgoing to mobile?


On Wed, Jul 30, 2014 at 4:51 AM, A J Stiles asterisk_l...@earthshod.co.uk
wrote:

 I'm having a problem with a new SIP trunk.

 Calls within the UK to fixed lines are fine, but calls to mobiles have
 noticeably poorer audio quality.

 I thought it might have been a codec issue; we have used G.726 for internal
 and external calls  (over primary ISDN and GSM).  So I tried allowing
 alaw,
 (G.711 A-law)  which is the native codec used within the PSTN in this
 country,
 but this made no improvement.

 We had
   disallow=all
   allow=g726

 in the [general] section of sip.conf.  In the section for one of the
 phones, I
 added
   allow=alaw
 and then inserted
   Set(SIP_CODEC=alaw)
 in the relevant part of extensions.conf.  For good measure, I also added
   NoOp(Codec was ${SIP_CODEC})
 in the h extension.  The messages in the Asterisk CLI appeared to show
 that
 the audio codec was correctly being set to alaw, and on hangup I got
 Codec
 was alaw, but there was no improvement to the sound quality.

 Is there something I am doing wrong, or do I need to get in touch with our
 SIP
 trunk provider?

 --
 AJS

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

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Re: [asterisk-users] Interesting new hack attack

[James Sharp]

On 5/22/2014 12:41 PM, Steve Murphy wrote:


So, these defenses can be employed to stop/ameliorate such
hacking efforts:

1. Keep your phones behind a firewall. Travellers, beware!
Never leave the default login info of the phone at default!
2. Never use the default provisioning URL for the phone,
with it's default URL or password.
3. Use fail2ban, ossec, whatever to stymie any brute force
mac address searches.
4. Use your firewalls to restrict IP's that can access web,
ftp, etc, for provisioning to just those IP's needed to allow
your phones to provision.
5. Keep your logs for a couple years.
6. Change your phone SIP acct passwords now, if you haven't
implemented the above precautions yet.


If I missed a previous post on this, forgive me.
Just thought you-all might appreciate a heads-up.


Encrypt your provisioning system if the phone supports it.  I had a 
cable/voip service provider who HTTPS provisioned by MAC without 
encryption and the provisioning URL was stored, unlocked, in the ATA. 
Had I been slightly more nefarious, I could have walked the the 
provisioning tree nice and slow and easily grabbed everyone's SIP 
credentials in the clear.


No hacking or cracking was involved.  The ATA doubled as the NAT router 
they handed out and gave the admin password out freely.


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[asterisk-users] FollowMe reinvites

[James Cloos]

For a sip-only application, what exactly is required to ensure that
calls completed via followme are reinvited?  Can it at all?

The code after outbound = findmeexec(targs, chan) calls ast_bridge_
call().  I don't see anything there which can cause a reinvite, yes?

When the same peer is used for both the incoming and outgoing legs,
it is a bit of a waste to proxy the rtp.

And even when the legs are associated with different remotes, I'd prefer
to proxy only when NATs a/or v4-v6 gatewaying are involved.

-JimC
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Re: [asterisk-users] new install: no re-invite and unwanted transcoding

[James Thomas]

I can't help on the first issue, but for the second have you tried doing
Set(${SIP_CODEC}=ulaw) before dialing the trunk? I'm in a similar situation
where we have g722 internally but our trunk provider only offers ulaw so I
see g722-slin-ulaw transcoding. I'm thinking of trying it here (on
1.8.14.1) to troubleshoot our occasional outbound issues.
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Re: [asterisk-users] CBAnn channel not going away in Asterisk 12

[James Sharp]

On 5/1/2014 10:38 AM, Richard Kenner wrote:

Please go ahead and open an issue and attach the refs log and the full DEBUG
log. That will allow us to understand what's occurring here.


I need to wait until I'm sure this isn't something I caused somehow,
so I need to first understand why I'm seeing this and nobody else is.



I had seen it as well but just chalked it up to not grokking how the 
CBAnn channels worked.



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Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[James Cloos]

 BF == Barry Flanagan barryf-li...@flanagan.ie writes:

BF Something like:

BF [peer_inbound]
BF context=peercontext
BF type=peer
BF host=192.168.1.1

BF ...should do the job.

That of course was something I tested, although I used the hostname;
perhaps the dual-v4/v6 got in the way?  

-JimC
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Re: [asterisk-users] srtp/dtls when sip is clear over lo

[James Cloos]

 JColp == Joshua Colp jc...@digium.com writes:

 Are you saying that asterisk doesn't care whether the sip is secure and
 will happily negotiate srtp depending only on whether the remote is
 willing to do so?  (That may come off as harsh; I do not mean it to be
 so, since it is what I want. :)

JColp Yes.

Thanks!

-JimC
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Re: [asterisk-users] srtp/dtls when sip is clear over lo

[James Cloos]

 JColp == Joshua Colp jc...@digium.com writes:

JColp The media is not carried over the SIP signaling,

Please give some credit, eh?

Given the sdp-negotiated srtp is not secure unless the sip is carried
over tls, the Best Practice is to require tls (or even sips: uris) to
agree to srtp.

Are you saying that asterisk doesn't care whether the sip is secure and
will happily negotiate srtp depending only on whether the remote is
willing to do so?  (That may come off as harsh; I do not mean it to be
so, since it is what I want. :)

And does anyone here have any operational experience on the matter of
what other endpoints are willing to do in such cases?

Thanks,

-JimC
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Re: [asterisk-users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

[James Cloos]

And related thereto:

What needs to be done on kama and ast to ensure that all incoming calls
which route through a given kama box always matches a sip.conf [section]
based on the socket(7)'s remote address, w/o any consideration of the
INVITE's sip headers or body?

I tried a several variations, but nothing quite worked.

-JimC
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[asterisk-users] srtp/dtls when sip is clear over lo

[James Cloos]

Given a box with a sip proxy listen(2)ing on 0.0.0.0 and chan_sip or
chan_pjsip listen(2)ing on 127.0.0.1, with ast sending rtp directly,
will ast negotiate srtp or dtls even ast and the proxy speak sip in
the clear over the lo interface?

Avoiding encryption over lo can aid debugging, but will doing so also
block secure media?

-JimC
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Re: [asterisk-users] Open Source Asterisk Polling Solution

[James Sharp]

On 4/23/2014 12:20 AM, Nick Cameo wrote:


That's about as simple as it gets.

A call file that goes to the dialplan.

A dialplan that consists of Read (which would play the message)
followed a GotoIf into a mailbox (either voicemail or Dial() to an
external number).

One hint for doing unattended dialing like this, make sure you're
dialing using a SIP or other digital method rather than, say, out an
analogue port that doesn't have decent answer detect.

And you can't just dump a whole bunch of call files into the system
at once, you'll need to meter them out based on the number of
concurrent outbound calls your provider will allow.


Hello James,

Good to see you here, and thank you very much. Though my basic idea of
how it will look using call files and dialplan is like what you and
others on here have pointed out. Yes,
we are using SIP for both origination and termination (just helping my
friend use some of our accounts used for PBX, and prepaid). I have been
using * for many years now however,
never for call center/predictive dialer type processes. Once I have got
this thing to call out and get calls coming in. It would be nice to
write to a database all the users that press
option on. I have a strong Java, PHP and SQL background. Will probably
need to make a call using AGI or such?

N.




You can go AGI, but there are direct ODBC handles available in the 
dialplan if you build Asterisk properly with the ODBC resources enabled. 
 That'd my personal preference from a performance standpoint.




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Re: [asterisk-users] Open Source Asterisk Polling Solution

[James Sharp]

On 4/22/2014 5:54 PM, Nick Cameo wrote:

Hello Everyone,

Thank you all for your response. The people I am doing it for run a
non-profit charity,
and are legally able to reach out to their customers. I will wire it
up to the DNC
however, for starters, I would like to get asterisk to:

i) Iterate through a list of numbers
ii) Play a pre-recorded message asking if they have waste they need picked up
iii) If they press one, forward the call to mailbox


That's about as simple as it gets.

A call file that goes to the dialplan.

A dialplan that consists of Read (which would play the message) followed 
a GotoIf into a mailbox (either voicemail or Dial() to an external number).


One hint for doing unattended dialing like this, make sure you're 
dialing using a SIP or other digital method rather than, say, out an 
analogue port that doesn't have decent answer detect.


And you can't just dump a whole bunch of call files into the system at 
once, you'll need to meter them out based on the number of concurrent 
outbound calls your provider will allow.



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Re: [asterisk-users] Open Source Asterisk Polling Solution

[James Sharp]

On 4/21/2014 1:47 PM, Mitul Limbani wrote:

Use vicidial for achieving the same.



Or call files (or AMI originate), a short bit of dialplan logic, and 
maybe a call to Queue().




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Re: [asterisk-users] Open Source Asterisk Polling Solution

[James Sharp]

On 4/21/2014 3:58 PM, Nick Cameo wrote:

On Mon, Apr 21, 2014 at 2:01 PM, James Sharp ja...@fivecats.org
mailto:ja...@fivecats.org wrote:

On 4/21/2014 1:47 PM, Mitul Limbani wrote:

Use vicidial for achieving the same.


Or call files (or AMI originate), a short bit of dialplan logic, and
maybe a call to Queue().




This is a nice and easy solution however, I do not know where to begin.
Can you gents kindly
elaborate or point us to the right directions (ie, howto tutorials)




Asterisk call files:

http://www.microalcarria.com/descargas/documentos/Linux/varios/Asterisk/asteriskdocs-docbook/docs-html/x1512.html

(Replace Channel with dial commands to your telephony provider, change 
the context and extension with an appropriate).


Dialplan:

You'll probably just need some Playback and Read commands to play your 
message and get the digit response.


Agents and Queues:

http://www.voip-info.org/wiki/view/Asterisk+call+queues

You'll probably want to use static/fixed agents unless you have a whole 
bunch of agents logging in and out.





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Re: [asterisk-users] IAXModem or T38Modem?

[James Cloos]

 j == joakimsen  joakim...@gmail.com writes:

j I wouldn't mind if someone posted on the list a known working provider
j with the proper configuration to use T.38. In my case I don't consider
j it an issue with the provider because they sent the proper T.38
j Invite, but Asterisk IMO does not know how to handle it.

Are you using a single credential-tuple with the provider?

If the provider supports T.38 and if you can separate out fax lines,
there is no need to stick asterisk between them and t38modem.  Just have
t38modem access the provider directly.  Hylafax will handle the rest.

(Look for things like sub-account, peer and/or trunk configs.)

-JimC
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Re: [asterisk-users] Numbers hackers call

[James Sharp]

On 3/26/2014 12:20 PM, Michelle Dupuis wrote:

If this is to 972 area code then the next digits should be 0X or 0XX but
they are not.  This differs from what I found documented for that area
code - I thought someone from the region might add to the discussion.
  Not sure if this reflected a premium service etc.  (But someone jumped
in with an explanation)


0X or 0XX is only if you're in country and need to dial with the 0 
national trunk code (much like dialing 1+ in the US for an in country 
but long distance call).  Someone dialing from outside the country 
doesn't need to add the zero, so they just use the 972 country code + 59 
prefix.



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Re: [asterisk-users] IAXModem or T38Modem?

[James B. Byrne]

On Mon, March 24, 2014 01:41, Mike Diehl wrote:
 Hi all,

 I'm installing Hylafax on my Asterisk system.  From what I've read, I can
 either use IAXModem or T38Modem to provide the virtual fax device.  So at
 the risk of starting a religious war, which one should I use?

 I don't mind running IAX if I have to.  I want as much flexibility and
 stability as I can get.

 So, what are your recommendations?

 Mike.


We use IAXModem-1.2.0 built from source and packaged as an rpm using
mock/rpmbuild together with Hylafax+-5.5.3 from epel.  Since April 2013 this
combination has been running two dedicated POTS lines through a TDM800-p8 on
our Atom CentOS-6.3 based Asterisk-11.7.0 (current version) box without any
reported difficulties (once I sorted out the upstart stanzas).

As this is the only combination I have experience with it is the only one I
can recommend.  But it has proven very reliable so far as I am aware and I
would be made aware pretty quickly if it was not.

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9 Brockley Drive  vox: +1 905 561 1241
Hamilton, Ontario fax: +1 905 561 0757
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[asterisk-users] Which is more efficient for 1 to many broadcasting?

[James Sharp]

Putting a whole bunch of people into a listen-only/muted Confbridge 
conference or getting the broadcaster audio into a MOH class and then 
just having callers attach to that MOH class?


Does the the muted side of a Confbridge Room still try to mix in audio 
from the muted channels or does it just disregard those channels and 
only run mixes against unmuted channels?


Now, if the answer is MOH is more efficient, can someone suggest a way 
for a channel to be the source of a MOH class?


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Re: [asterisk-users] Which is more efficient for 1 to many broadcasting?

[James Sharp]

On 3/18/2014 6:58 PM, Paul Belanger wrote:

On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote:

Putting a whole bunch of people into a listen-only/muted Confbridge
conference or getting the broadcaster audio into a MOH class and then just
having callers attach to that MOH class?

Does the the muted side of a Confbridge Room still try to mix in audio from
the muted channels or does it just disregard those channels and only run
mixes against unmuted channels?

Now, if the answer is MOH is more efficient, can someone suggest a way for
a channel to be the source of a MOH class?


What sort of channel count are you looking for? We did some load
testing recently and found less people in a bridge is better then
more.  Audio source location didn't really matter much.



A few hundred to start with, but as with everything, I'd like to scale 
up as far as I can.  And, of course, it makes sense that less people in 
a bridge is better than more but that's not quite what I'm asking.


Is it more efficient to have, for example, 701 people in a confbridge 
room (700 muted users + 1 person yapping) or to have 700 people dialed 
in and just running the MusicOnHold application with said person yapping 
away via some audio source.



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Re: [asterisk-users] Replying to Posts

[James B. Byrne]

On Thu, March 13, 2014 15:32, Kevin Larsen wrote:
 On 13/3/14 6:27 pm, Eric Wieling wrote:
  This is an example of why I top post.   Who wrote what?


+1-1 = 0

I do not care about where people put their replies so long as I can figure out
who is answering what.  What I do not like to read is this interminable
religious dogma about the 'natural' order of writing.  This is the second or
third list this week in which this B.S. has shown up in my inbox.

In written business communication, in contrast to tech-speak customarily found
on mailing lists, ones answer always goes before any quoted context.  Not
because it has to, it is just that I have seldom, if ever, seen it done any
other way. And regular business communication with non-technical folk
comprises well over 75% of my daily written communication.

And while I understand the cultural motivation behind the dogma of bottom
posting I remain sceptical respecting its utility.  Is there any objective
evidence whatsoever that top or bottom posting makes any difference to the
reader's understanding of the message?  Does any rigorously determined data
exist to support that contention?  If not then this is simply a matter of
trying to impose a set of arbitrary cultural values cloaked in the guise of
technical superiority.


 Of course, if you use a mail client that's capable of quoting correctly,

 it all works beautifully.


 Outlook can quote correctly, but it is an all or nothing setting it would
 appear. Lotus Notes actually handles it better as there is a Reply option
 for normal email and a Reply With Internet-Style History that I use for
 this list. I don't have any problems following the rules of the list, but
 I am fully on the side of the Replies should go at the top group and
 would vote for a change in the rules.



And do not even start on the Chevy vs. Ford debate respecting the technical
superiority of Pine over Outlook.  GAWD... Life its too short as it is.


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Hamilton, Ontario fax: +1 905 561 0757
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Re: [asterisk-users] Host = Dynamic in a Register Free Setup

[James Sharp]

On 2/18/2014 2:09 PM, Eric Wieling wrote:

No.  Asterisk will accept calls from unregistered devices, but you have to 
enable guests I sip.conf and hope your dialplan is secure.  No sane person does 
this.

Asterisk cannot send calls to a device unless it knows the address from a 
register or from a host= entry for the peer.

You may not like it, but this is the way Asteirsk has worked for the past 15 
years.


Isn't there also autocreatepeer?



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[asterisk-users] Looking for some guidance with the Asterisk 12 ARI/API

[James Wystead]

Hi - I figured this was probably the best place to ask this question

Is there anyone that has done anything practical with the API and/or Real
Time Database config?

If so, I would like to pick your brains if I may.

Thanks - G
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Re: [asterisk-users] answering machine screening with MixMonitor

[James Sharp]

On 2/5/2014 12:09 PM, G. Paul Ziemba wrote:

I'm using asterisk 1.8 as an answering machine. I'd like to
hear the calls it answers aloud in case I want to pick up and
interrupt the call.

There are a few articles describing, for example, three-way
calling a monitor phone set to auto-answer, but I couldn't
find anything that described how to just send the audio to
a local speaker.


A local speaker connected to the Asterisk box itself?  Console channel 
driver, chan_alsa (or chan_oss for old drivers).


You'll probably end up with kind of a Rube Goldbergish approach, 
probably something involving ChanSpy or a conferencebridge to take the 
place of mixmonitor.





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[asterisk-users] Asterisk 11 - looking for ideas and a possible solution

[James Wystead]

First, let me thank all of you for your input - I was looking for some sort
of an API to interface with Asterisk via REST. Python, Ruby or something
that we could webify using PHP.

Some of you suggested Asterisk 12 - I love the idea, but unfortunately, we
have an Asterisk 11 install. It seems, and I could be wrong, that Asterisk
12 is a whole new 'ball game' so to speak. The big thing is pjsip - nice
idea and I really like it. It looks like a bit of a learning curve and I
don't want have to re-learn at this point - I will, but not right this
second!  However, I have questions:

1. Is it necessary to use pjsip for Asterisk 12 as the extension.conf,
sip.conf has changed gears a little. Can I backtrack to the previous SIP
stack, which segways into another question:

2.  If this is possible (the above scenario) is it still possible to use
these new cool features along with the previous SIP stack so that I can
implement the API?

Thanks to anyone who can give me insight. I don't mean to sound lazy, but
v12 seems to be so new that the information is not as plentiful as previous
versions. I figured there has to be someone that was 'on the ball' with
this and would be happy to share their knowledge!

Thanks again!

G
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Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution

[James Wystead]

Oh - no kidding!

What we want to do is be able to create users, voicemail accounts and some
of the basic features. Nothing fancy, I'll create the dialplan by hand of
course.

So, if I understand, I can use the chan_sip, using the old ways (Asterisk
11) and the Asterisk 11 CLI commands - that is what I'm familiar with.

I can use this and still use the new AMI/API features as well as the real
time database config?

I just want to be 100% clear - sometimes I need to hear an explanation a
couple of times before it sinks in!

Thanks


On Sat, Jan 25, 2014 at 9:07 AM, Joshua Colp jc...@digium.com wrote:


 On 14-01-25 09:54 AM, James Wystead wrote:
  First, let me thank all of you for your input - I was looking for some
  sort of an API to interface with Asterisk via REST. Python, Ruby or
  something that we could webify using PHP.

 It all depends on what you want to do the with API as well...

  Some of you suggested Asterisk 12 - I love the idea, but unfortunately,
  we have an Asterisk 11 install. It seems, and I could be wrong, that
  Asterisk 12 is a whole new 'ball game' so to speak. The big thing is
  pjsip - nice idea and I really like it. It looks like a bit of a
  learning curve and I don't want have to re-learn at this point - I will,
  but not right this second!  However, I have questions:
 
  1. Is it necessary to use pjsip for Asterisk 12 as the extension.conf,
  sip.conf has changed gears a little. Can I backtrack to the previous SIP
  stack, which segways into another question:

 The chan_sip module has not been removed and works perfectly fine in 12.
 You can even use both, provided you bind each to different ports.
 Dialplan is also the same as in the past.

  2.  If this is possible (the above scenario) is it still possible to use
  these new cool features along with the previous SIP stack so that I can
  implement the API?

 Depends on what you mean by cool new features. The ARI (Asterisk REST
 interface) work is agnostic of channel drivers, it has no specialized
 logic for any and any present channel driver can be used with it.

 Cheers,

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org

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Re: [asterisk-users] Asterisk 11 - looking for ideas and a possible solution

[James Wystead]

On that note, Joshua , while I have your attention. I wanted to ask this:

Is there an available API or some other medium that can interface with
Asterisk? Something that we can write, perhaps some PHP scripts and put and
get commands to some API that can manage Asterisk in that way?

I also understand that there is a real time database module.

Thanks - G


On Sat, Jan 25, 2014 at 9:47 AM, Joshua Colp jc...@digium.com wrote:

 On 14-01-25 10:44 AM, James Wystead wrote:
  Oh - no kidding!
 
  What we want to do is be able to create users, voicemail accounts and
  some of the basic features. Nothing fancy, I'll create the dialplan by
  hand of course.

 The Asterisk REST interface does not currently provide this
 functionality. It's not for management, it's for writing telephony
 applications (such as a new app_queue or app_voicemail) outside of
 Asterisk.

  So, if I understand, I can use the chan_sip, using the old ways
  (Asterisk 11) and the Asterisk 11 CLI commands - that is what I'm
  familiar with.

 Yes.

  I can use this and still use the new AMI/API features as well as the
  real time database config?

 Yes.

 Cheers,

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org

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