Re: [asterisk-users] Question about Sip Trunks who support Stir Shaken

[Jeff LaCoursiere]

Bandwidth.com, although there are minimums to meet.

Cheers,

Jeff LaCoursiere
StratusTalk, Inc.

On Fri, Aug 18, 2023 at 7:52 AM TTT  wrote:

> Check out Twilio
>
>
>
>
>
> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Behalf Of *Federico
> *Sent:* Thursday, August 17, 2023 11:49 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' <
> asterisk-users@lists.digium.com>
> *Subject:* [asterisk-users] Question about Sip Trunks who support Stir
> Shaken
>
>
>
> I am looking for a decent provider of SIP Trunks but it has to pass the
> Stir Shaken token to the next carrier. Does anybody know about any?
> Sipstation from Sangoma, does not support Stir Shaken. ( Case #01466843 /
> 001300G8PLG / MAIN / Open [ ref:_00D306mPe._5004U1BlBLF:ref ])
>
> Although it’s mandatory, somehow they think it’s ok. Go figure.
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Re: [asterisk-users] Get channel variables via ARI/AMI

[Jeff LaCoursiere]

I need to get hooked up with this class, I could have students doing 
projects for homework :) Interested in RTCP?


j

On 6/26/23 7:45 PM, TTT wrote:


I’m in training, so I have to demonstrate something SIP related.  I 
figure it would be cool to hack a call, hanging it up while in 
progress from outside Asterisk.  Doing so will demonstrate 
use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc.


Practical value: *zero*

J

Who knows, maybe this will have an actual application for someone 
someday.  In practical terms I think building a proxy would be the 
right way to manipulate the SIP for a call in progress, but that 
sounds like a huge project.  I’ve got to demonstrate something by end 
of week.


*From:*asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] 
*On Behalf Of *Jeff LaCoursiere

*Sent:* Monday, June 26, 2023 6:20 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Get channel variables via ARI/AMI

On 6/26/23 9:00 AM, Joshua C. Colp wrote:

On Mon, Jun 26, 2023 at 10:57 AM TTT  wrote:

I am connecting to the ARI with subscribe all, so I can see
channels being created.  I now want to extract a variety of
header variables (at the moment the from and to tag).  I tried
to read them from the ARI but Asterisk refuses since the
channel is not in a  stasis app.

Is there a way to read these from either the ARI or AMI ?  I’m
trying not to modify the dialplan.

ARI, No.

AMI, Yes[1].

[1]
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar

I'm curious what the actual application is here - you want to connect 
to AMI to pull information that you will use to pretend to be a leg, 
just to send "BYE", when you could just hangup the leg with AMI (or do 
just about anything else you might think of).  Sometimes it is better 
to fully explain what you are trying to accomplish, and some folks 
here can try to steer you towards a workable solution.  It almost 
sounds... nefarious.


Cheers,

--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell



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Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
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Re: [asterisk-users] Get channel variables via ARI/AMI

[Jeff LaCoursiere]

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote:

On 6/26/23 9:00 AM, Joshua C. Colp wrote:

On Mon, Jun 26, 2023 at 10:57 AM TTT  wrote:

I am connecting to the ARI with subscribe all, so I can see
channels being created.  I now want to extract a variety of
header variables (at the moment the from and to tag).  I tried to
read them from the ARI but Asterisk refuses since the channel is
not in a  stasis app.

Is there a way to read these from either the ARI or AMI ?  I’m
trying not to modify the dialplan.


ARI, No.
AMI, Yes[1].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar


I'm curious what the actual application is here - you want to connect 
to AMI to pull information that you will use to pretend to be a leg, 
just to send "BYE", when you could just hangup the leg with AMI (or do 
just about anything else you might think of).  Sometimes it is better 
to fully explain what you are trying to accomplish, and some folks 
here can try to steer you towards a workable solution.  It almost 
sounds... nefarious.



Meant that towards TTT, not Josh, in case that wasn't clear.

--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
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Re: [asterisk-users] Get channel variables via ARI/AMI

[Jeff LaCoursiere]

On 6/26/23 9:00 AM, Joshua C. Colp wrote:

On Mon, Jun 26, 2023 at 10:57 AM TTT  wrote:

I am connecting to the ARI with subscribe all, so I can see
channels being created. I now want to extract a variety of header
variables (at the moment the from and to tag).  I tried to read
them from the ARI but Asterisk refuses since the channel is not in
a  stasis app.

Is there a way to read these from either the ARI or AMI ?  I’m
trying not to modify the dialplan.


ARI, No.
AMI, Yes[1].

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+ManagerAction_Getvar


I'm curious what the actual application is here - you want to connect to 
AMI to pull information that you will use to pretend to be a leg, just 
to send "BYE", when you could just hangup the leg with AMI (or do just 
about anything else you might think of). Sometimes it is better to fully 
explain what you are trying to accomplish, and some folks here can try 
to steer you towards a workable solution.  It almost sounds... nefarious.


Cheers,

--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
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[asterisk-users] PMS integration

[Jeff LaCoursiere]

Howdy,

Has anyone worked on a Mitel-2000 emulation for PMS integration (Hotel 
mgmt systems)?  Hoping to get my hands on the protocol definition 
(RS-232!!) for check-in/check-out/housekeeping/CDR, but if someone has 
already done I would totally buy it.


Cheers,

--
Jeff LaCoursiere
StratusTalk, Inc.


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Re: [asterisk-users] Intro and question

[Jeff LaCoursiere]

If you just want something easy to use out of the box, install the 
FreePBX distro.


Cheers,

j

On 4/6/23 11:08 AM, Steve Matzura wrote:

Anthony,


No, I had no intention of doing any of those things, for I know not 
what they are or why I would need or want to be doing them. Maybe I 
should have just stuck with the original idea of installing from 
Debian distro. I'm exploring the phreaknet option now. If I come up 
with a running system, I'm just going to leave it that way and work 
with it as it is.



On 4/6/2023 10:35 AM, Antony Stone wrote:

On Thursday 06 April 2023 at 15:48:24, Steve Matzura wrote:


this is the first time I have attempted a
from-scratch installation and setup on my own.

..


Then the weeds started to appear, and I was off into them.

The first was the mention of Alembic.
Reading on, I found this, regarding an SQL database:
SQL? Database? Where ... what ...
Thanks in advance for any assistance.
Well, my first question would be "are you intending to use Asterisk 
Realtime
features (ie: configurations in database tables instead of text 
files) in this

installation?"

If you are, then you do need to install a few more packages on your 
Debian
system, but if not, then there is no reason to pay any attention at 
all to

anything to do with Alembic, Realtime, SQL etc.


Antony.





--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell


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Re: [asterisk-users] github - mlan

[Jeff LaCoursiere]

Ah, great advice, thanks!

j

On Tue, Feb 7, 2023 at 10:09 PM John Runyon  wrote:

> If you clone one of their repo's you can see their email address in the
> commit log...
>
> On Tue, 7 Feb 2023 at 16:56, Jeff LaCoursiere 
> wrote:
>
>> Hi all,
>>
>> Curious if the github user "mlan" is on this list?  Could you please
>> contact me off list if so, I was hoping to reference your work in a talk
>> at Astricon next week, and... I don't know how to contact github users
>> lol.
>>
>> Cheers,
>>
>> --
>> Jeff LaCoursiere
>> StratusTalk, Inc.
>> 703 496 4990 x108
>> 815 546 6599 cell
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
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[asterisk-users] github - mlan

[Jeff LaCoursiere]

Hi all,

Curious if the github user "mlan" is on this list?  Could you please 
contact me off list if so, I was hoping to reference your work in a talk 
at Astricon next week, and... I don't know how to contact github users lol.


Cheers,

--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell


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Re: [asterisk-users] Run asterisk -rx "command" and get plain text output

[Jeff LaCoursiere]

Haven't tried this, but try piping through 'strings'

j

On Wed, Aug 3, 2022 at 6:13 PM Carlos Chavez  wrote:

> The "-n" option only works on startup and cannot be used when Asterisk
> is already running (I tried and I get an error).  We are using version
> 18.12.1.  The output I want to capture is:
>
> asterisk -rx "queue show"
>
> If I capture the output to a file or straight to the command line I
> get the colorized output.
>
> I found a way to strip the ANSI codes using SED:
>
> asterisk -rx "queue show" | sed -e 's/\x1b\[[0-9;]*m//g'
>
>
> On 8/3/2022 10:35 AM, Joel Serrano wrote:
>
> Have you tried adding “-n”?
>
> Also, what version of asterisk are you using? newer versions only have
> colorized output when your are connected to the console (-r) not for remote
> commands (-rx)
>
>
>
> On Wed, Aug 3, 2022 at 08:21 Carlos Chavez  wrote:
>
>>  I usually like to have the colorized output when looking at
>> asterisk output but I need to get some info by running "asterisk -rx"
>> and get just plain text output so I can mail it.  Right now I get ANSI
>> codes in the output.  Is there a way to get plain text output for just
>> that script and not disable colors for everything?
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> +52 (55)8116-9161
>>
>>
>> --
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>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
>
> --
> _
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> Check out the new Asterisk community forum at:
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>
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Re: [asterisk-users] Delay when dialing...

[Jeff LaCoursiere]

Are you sure the call has been sent? Some phones have odd dialplans 
installed, and may not send the call to the SIP relay until you meet the 
dialplan reqs, press #, or otherwise wait the inter-digit timeout before 
the call is actually placed.


If this is the case you need to take a hard look at the dialplan string 
for the endpoint, and edit accordingly.


j

On 7/23/21 9:35 AM, Carlos Chavez wrote:
Thank you.  The server is running dnsmasq locally for DNS resolution 
and all queries resolve properly.  I just added the hostname to 
/etc/hosts and restarted but the delay persists.


On 7/23/2021 1:41 AM, Jean Aunis wrote:

Le 22/07/2021 à 18:32, Carlos Chavez a écrit :
    I started noticing a few days ago that whenever I dial any 
number or extension there is a delay of 5 to 10 seconds before 
Asterisk reacts.  I see nothing on the CLI for that time and then 
the call goes through.  I have checked my network to make sure there 
is nothing slowing down packets between the phones and the server.


    Any settings I should check on the Asterisk side?  This is 
happening with all phones (several brands).



Hi,

I've seen this problem several times when there is no DNS resolution 
of Asterisk's hostname.


Try to add your hostname to /etc/hosts and check if it's better.

Regards,

Jean





--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell


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Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

The "inbound piece" is "what do I do with the tag information"?

Should I find a way to present the fact that a call has an A rating?

Should I offer to block calls with a C rating?

It would be great to see asterisk be able to unpack this stuff and have 
it available as a dialplan variable and in the CDRs.


Jeff LaCoursiere
StratusTalk, Inc.

On 3/11/21 6:21 PM, Alexander Perkins wrote:
Hi Jeff.  What exactly do you mean by the 'inbound piece'?  I've spent 
quite a lot of time with the folks at TILTX understanding the 
framework; but I am not exactly sure what you mean by the 'inbound piece.


Greg/Doug, like many folks here, we use LCR.  So, the terminating 
carrier is not necessarily the one that issued us the telephone 
numbers.  So, they will not sign it or simply cannot sign it.  
Remember that a very limited number of companies can actually sign the 
calls; the rest have to buy it from these 'Service Providers'.


And there is another situation - the company you purchase your numbers 
from and the company you place your calls through may be different and 
both may not be able to sign your calls. Again, a very limited number 
of service providers that can actually sign your calls.  So what do 
you do in that scenario?  You have to find a Service Provider that can:


1.  Verify you own that telephone number(s).
2.  Sign your calls.
3.  Provide you with the technical means to do so.

So, that's that...  I hope this makes sense.

Alex



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Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

To be honest, that is the logic we ended up with, and are dumping our 
LCR.  The savings aren't worth the headache.  We don't have 1M numbers, 
but we have a significant number.  We can't quite get down to one 
carrier (and don't really want to), but we can keep outbound calls on 
the carrier that "owns" them, and not worry about this.


Jeff LaCoursiere
StratusTalk, Inc.

On 3/11/21 8:12 PM, d...@donkelly.biz wrote:


You said it in your first post when you said “I reallt don’t 
understand.” You don’t understand the business that these people are 
in. A few people showed you a few examples of why it’s important to 
use more than one carrier--and there are other reasons that 
stir/shaken is a big deal for some of us.


It clearly isn’t a big deal for you, so you probably don’t have much 
to add to the discussion.


--Don

*From:* asterisk-users  *On 
Behalf Of *Sebastian Nielsen

*Sent:* Thursday, March 11, 2021 7:21 PM
*To:* 'Mailing List' 
*Subject:* Re: [asterisk-users] STIR/SHAKEN

1:  1M DID’s? Then I would go straight out and say you are a phone 
operator, and then getting your own STIR/SHAKEN certificate shouldn’t 
be a problem at all. Thats a massive amount of numbers, 
unrealistically many numbers for any company ever except for those 
that are a phone operator.


2: For me, its seems like hunting for nano-cents. I checked around 
when I got my DID and call account for my own personal use, and the 
prices aren’t that different. Its really not worth the effort for what 
you save. Checked with several operators and the prices are almost the 
same per minute, its like one operator has like 0.016 per minute and 
another has 0.014 … not gonna save much on that. Might save like 1$-2$ 
per month on choosing the latter operator.


3: Why? Consolidiate all your agreements to 1 single operator that 
handles everything, and everything will be so much simpler. Then you 
are simply a trunk ccustomer to that particular operator, no need to 
handle all this with signing and certificates and everything..


To save a little tiny nano-cent from each minute of call..

*Från:*asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> 
<mailto:asterisk-users-boun...@lists.digium.com>> *För *Joel Serrano

*Skickat:* den 12 mars 2021 01:52
*Till:* Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>

*Ämne:* Re: [asterisk-users] STIR/SHAKEN

Hi,

I wanted to add some comments to Sebastian's response:

1- When you have a lot of DIDs, you can't just "port" them over from 
company1 to company2. Try to have 1M or so DIDs and ask if you can 
just port them. No no, not that simple. There is a process that a lot 
of times is not worth the cost/risk/etc.


2- What happens if company1 has very good pricing for DIDs, but 
extremely high rates for placing outbound calls, and company2 has 
super aggressive pricing for the destinations you use most, but sells 
DIDs very expensive? Mix and match? :)


3- What do you do, when instead of having 1 outbound carrier, you have 
several 50?


At the end I think you are mistakenly comparing apples to oranges, 
your DID provider has nothing to do with your outbound carrier, can 
the DID provider also give you outbound calling? Most likely, but that 
doesn't mean that the best way to go is to route outbound calls via 
the carrier that is providing you DIDs.


On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen <mailto:sebast...@sebbe.eu>> wrote:


I reallt don’t understand why people simply use the same operator
to terminate your calls, which also provide DIDs for you.

Then you don’t need to touch this at all, your carrier will do all
the STIR/SHAKEN handling for you, you are just a PBX customer.

And then the operator then simply limits your account to only
present your DID as outgoing number.

Seems to be a unneccesary complicated solution just to have your
numbers at company 1 and have your call termination at company 2.

So fricking unneccessary.

What I know there is requirements of number portability, so as
long as company 2 can handle DIDs (ergo ”own” DIDs) you should be
able to move your DIDs from company 1 to company 2 – then company
2 owns your DIDs.

Best regards, Sebastian Nielsen

*Från:*asterisk-users-boun...@lists.digium.com
<mailto:asterisk-users-boun...@lists.digium.com>
mailto:asterisk-users-boun...@lists.digium.com>> *För *Alexander
Perkins
*Skickat:* den 12 mars 2021 01:23
*Till:* asterisk-users@lists.digium.com
<mailto:asterisk-users@lists.digium.com>
*Ämne:* Re: [asterisk-users] STIR/SHAKEN

Hi Jeff.  What exactly do you mean by the 'inbound piece'?  I've
spent quite a lot of time with the folks at TILTX understanding
the framework; but I am not exactly sure what you mean by the
'inbound piece.

Greg/Doug, like many folks h

Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

Hi Alex,

Are they doing anything on inbound for you, and have you made any 
decisions about how you will display the tag to your customers? I have 
been focusing on the outbound piece of this, just starting to think 
about what to do with the incoming data...


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.

On 3/7/21 7:08 PM, Alexander Perkins wrote:
Hi Greg.  In our use case, we purchase DIDs from them.  So, they are 
the inbound carrier (they are a CLEC and IPES) and STIR/SHAKEN Service 
Provider.  However, we do not use them for termination.  They offer 
service termination, but we do not use them due to 
existing contracts.  So, in order to have our calls signed, we needed 
them to do it.  The biggest issue we've come across is the number of 
companies /able to /provide this service is limited, especially to the 
Asterisk community. I stress able to because even though some 
companies are Service Providers, they are simply not 
technically capable of offering it.


I will send you my contact's information at TILTX privately.  He's a 
subject-matter expert with the STIR/SHAKEN framework and he's offered 
us invaluable help.


Thanks,
Alex

On Sun, Mar 7, 2021 at 1:43 PM Greg Troxel <mailto:g...@lexort.com>> wrote:



Alexander Perkins mailto:alexanderhenryperk...@gmail.com>> writes:

> They ended up creating an AGI script for us that handles
everything.  At
> the end of the day, all we needed to do was pull down the
script, and add
> the exten => s,n,AGI(TILTX-SHAKEN.agi) command and it handles
> everything else.

I wonder if you could step back and explain the big picture, as
I'm not
really following this.   As I understand it:

  usually asterisk is used as a pbx

  STIR/SHAKEN is a protocol run between carriers to prove the
authority
  to use the claimed callerid

  when someone gets service from a carrier and connects to it from
  asterisk, I would expect the carrier to basically filter the claimed
  callerid to be from the set of values recorded with your account as
  legit, and for the carrier to do the STIR/SHAKEN authentication.

So I wonder if your asterisk instance is connecting to the PSTN as a
top-level carrier, or, more likely, I am confused in some way.




--


*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
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Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

On 3/7/21 2:26 PM, Doug Lytle wrote:

On 3/7/21 1:43 PM, Greg Troxel wrote:

So I wonder if your asterisk instance is connecting to the PSTN as a
top-level carrier, or, more likely, I am confused in some way.


Greg,

I think this is the case for quite alot of those here.

For me though, I just manage the on premise PBX and my carrier handles 
the STIR/SHAKEN part.


Doug



Hi,

There are issues for those of us that use multiple upstream carriers for 
call termination, with LCR for example.  If you send your calls out the 
same provider that supplies your inbound DID, your calls should get the 
"A" rating and your callers should have no issues. At present if I send 
calls out a provider that does NOT handle the DID in the caller ID 
field, it gets a "B" rating.  I don't think this will pose a problem for 
the forseeable future - I don't see carriers marking these as "spam",  
they just won't get the ultra-special "secure" mark.


Also good to note the upcoming deadline does NOT mean call blocking, 
just call tagging.  The blocking bit will be up to the end user, though 
I could see phones shipping with default settings that may do so.


Basically we can't do LCR anymore.  Outbound calls are locked to the 
provider that gave us the DID.  I'm not sure that's really a bad thing, 
its less headache than for us to try to become a signing authority.


I think the whole thing is still very fluid.  Didn't even mention call 
forwarding issues.


j

--


*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
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Re: [asterisk-users] AGI Script Returning 4

[Jeff LaCoursiere]

Out of RAM/swap?

j

On 1/30/21 1:18 PM, Alexander Perkins wrote:
HI All.  I have a really strange issue that I'm two months into 
troubleshooting; however, I cannot figure it out.  I have an AGI 
Script (PHP) that runs every time a call comes into my Asterisk box.  
Most of the time, it runs without any issue. However, every now and 
then, the PHP-AGI script fails after it is executed and simply returns 
'returning 4'.  I verify the PHP script begins to run.  However, it 
appears to just stop.  I have placed try/catch statements everywhere, 
but it does not seem to hit them.


Just to verify, this is the same script running over and over with the 
same parameter.


Any ideas/suggestions as of what can be happening?

Thanks,
Alex



--

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
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Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

On 1/28/21 2:08 AM, Alexander Perkins wrote:
Jeff, yes.  The process is long.  It is actually around one year.  We 
ended up going with a SHAKEN Service Provider named Technology 
Innovation Lab (www.tiltx.com <http://www.tiltx.com>). They have been 
awesome.  They are certified in Asterisk and catered the solution to 
our Asterisk install.  Highly recommend them. Their email for SHAKEN 
is 0...@tiltx.com <mailto:0...@tiltx.com>.


Anyways, give them a shot.  Took us a while to find a SHAKEN Service 
Provider that knew Asterisk.


Alex

Thanks Alex!  I'll give them a call.  I'm planning to make a big post in 
a week or so with all I have learned, hopefully will help others unsure 
where we stand.  June is coming up quick!


Cheers,

--


    *Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>



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[asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

On 1/25/21 12:12 PM, Steve Edwards wrote:

On Mon, 25 Jan 2021, Jeff LaCoursiere wrote:

So how does this guy get around it?  It sounds to me like he is 
offering to sign calls for whoever, which IMO totally defeats the 
purpose.


IIRC, back when he first started hawking his solution, he accepted 
everything. Numbers from Vitelity, my old out of service copper 
number, 555-555-.


I'm all for the discussion, but can you start a new thread so we don't 
keep associating the innocent party (the OP) with this spammer.


Excellent point, started new thread.

In my digging today it seems I need to become a SHAKEN service provider, 
and there is a rather lengthy and difficult process to go through to 
become one (this slide from a Bandwidth.com seminar):


Has anyone gone through this recently?  Does it really still take 7 to 9 
months?  That seems crazy.


Cheers,


--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell

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[asterisk-users] Fwd: Your message to asterisk-users awaits moderator approval

[Jeff LaCoursiere]

A 40Kb limit seems a bit draconian these days.  I simply attached a 
small pic to illustrate a point.  May I vote to up the limit?  100K?


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.

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STIR/SHAKEN

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Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)

[Jeff LaCoursiere]

Thanks Joshua, that was warranted IMO. I have to admit I am curious 
about his solution, though, and maybe we can turn this into a discussion 
about how we are handling STIR/SHAKEN collectively.


Our biggest issue is the fact that we use (our own) LCR to choose 
outbound routes for calls in real-time.  An outbound call could end up 
on any of several upstream providers, not necessarily the one that 
"owns" the DID that is making the outbound call.  Because of this we 
cannot rely on our upstreams to be the originating carriers - they won't 
be able to sign for calls from DID numbers they have no record of. We 
aren't alone in this, and Bandwidth.com is making noise about this issue 
and the "forward" problem, where we bounce a call to another number, and 
the original token info is lost.  I'm pretty sure this lands us in the 
"need to be an originating provider and generate crypt tokens".  I'm 
only just beginning diving into what that entails.


So how does this guy get around it?  It sounds to me like he is offering 
to sign calls for whoever, which IMO totally defeats the purpose.  So if 
I am some spammer and find my calls are rejected, I can just get him to 
sign them for me?  If I were him I would get a bunch of lawyers ready 
for when he becomes responsible for what they end up doing.  Isn't that 
the whole idea?


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.

On 1/25/21 7:44 AM, Joshua C. Colp wrote:
On Sun, Jan 24, 2021 at 6:50 PM Steve Edwards 
mailto:asterisk@sedwards.com>> wrote:


On Sun, 24 Jan 2021, Saint Michael wrote:

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

"I have a commercial solution that works for any version of
Asterisk, if
interested contact me at venefax at the Google mail service."

Fixed. If you're going to post a commercial solution on a
non-commercial
forum, at least be up front about it.


Greetings all,

Just so everyone is aware I previously gave a final warning on 
commercial solicitation on the asterisk-users mailing list to this 
individual[1]. As they've continued to do so I've removed them from 
the mailing list. You are free to communicate with them directly.


Cheers,

[1] 
http://lists.digium.com/pipermail/asterisk-users/2020-July/295208.html


--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com <http://www.sangoma.com/> and 
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Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell

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Re: [asterisk-users] Asterisk registrations - state?

[Jeff LaCoursiere]

On 12/15/20 3:19 PM, Antony Stone wrote:



There is no functionality present to have Asterisk SUBSCRIBE to upstream
servers, receive updates, and locally use them.

Hm, thanks for the clarification, this confirms what I suspected.

Can anyone suggest an alternative application I could sensibly use alongside
Asterisk in a production environment to achieve this (I'm specifically looking
for a way to maintain presence information for a Busy Lamp Field, and to get
notifications about Voicemail events).

I'm aware of both SIPp https://github.com/SIPp/sipp and SipSak
https://github.com/nils-ohlmeier/sipsak, however these are (to me at least)
testing scenario tools designed for "short term" use to place and receive
calls, rather than maintaining long-term subscriptions to SIP accounts,
waiting for notifications of events, so I'm somewhat wary about considering
them for use in a production environment (by which I mean I'm looking for
something which can subscribe to a few tens of SIP accounts (across several
servers) and get the event notifications for those accounts, over a period of
days and weeks, and pass them to my display application in almost any
reasonable form).


Any ideas out there?



It sounds to me like you just need a reasonable SIP softphone that can 
register with multiple SIP accounts (they all do) and show presence for 
the extension registered (I think most want you to pay for this feature 
versus the freebie client).


Why run asterisk?

Am I missing something?

Cheers,

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STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
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Re: [asterisk-users] which linux for asterisk?

[Jeff LaCoursiere]

Ya, I wouldn't say this is our normal behaviour - to complete ignore a 
running production system for four years, but if it ain't broke...


On 12/9/20 7:12 AM, Michel FACERIAS wrote:

Debian6 is very stable too.

I have some... in production :-)

Or not.

---
Michel FACERIAS

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http://www.facerias.org

Le 2020-12-09 14:06, Dmitry Melekhov a écrit :

09.12.2020 16:52, Jeff LaCoursiere пишет:



This machine I visited yesterday in our data center... it is running 
Ubuntu 14... I would say this is a pretty stable platform :)


Ubuntu 14...

It is not supported for years now.

This is not our method, we are replacing Centos 6 servers now...





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Re: [asterisk-users] which linux for asterisk?

[Jeff LaCoursiere]

This machine I visited yesterday in our data center... it is running 
Ubuntu 14... I would say this is a pretty stable platform :)


Cheers,

j

On 12/9/20 5:00 AM, Dmitry Melekhov wrote:

09.12.2020 13:20, Frank Vanoni пишет:

On Wed, 2020-12-09 at 11:03 +0400, Dmitry Melekhov wrote:


what is best choice ? Oracle? Ubuntu?

I'm running Asterisk since several years on Ubuntu without any issues.

Debian should be fine too.



Thank you.

This gives me just about 3-4 years of support, considering 2 years 
between LTS,


and upgrading remote server can be pain.

Anyway, this is good option...





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Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[Jeff LaCoursiere]

I didn't want to post this because its kind of ugly, but we *did* 
actually do it a number of years ago to get around this issue with chan_sip.


Our original architecture was based on LXC, and we had large servers 
running hundreds of containers, each running asterisk.  The "host" ran 
asterisk too, as the gateway for all the container instances.


We once used two of those containers to run asterisk on specific host 
interfaces (one instance bridged to one nic, the other to the other).  
The host asterisk would route calls out one container or the other, with 
the effect you are looking for...


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.


On 10/29/20 7:42 PM, David Cunningham wrote:

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a 
specific IP address for its end of the communication for a specific 
device? Something like:


[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There is the 
"extenip" setting, but it's really designed for NAT, and can only 
appear in the [general] section.


Any suggestions would be greatly appreciated.


On Sat, 24 Oct 2020 at 09:43, David Cunningham 
mailto:dcunning...@voisonics.com>> wrote:


OK, thank you George.


On Sat, 24 Oct 2020 at 03:16, George Joseph mailto:gjos...@digium.com>> wrote:



On Thu, Oct 22, 2020 at 4:13 PM David Cunningham
mailto:dcunning...@voisonics.com>>
wrote:

Hi George,

Thank you for the response. I'm a little unclear on what
you mean by a transport. We're using chan_sip, not pjsip.

Do you mean a device in sip.conf, using bindaddr to set
the address to bind for that device? We've only used
bindaddr in the [general] section before, but if it will
work in a device that could be the answer.


Sorry.  I just assume chan_pjsip these days.  Not sure how
you'd do it for chan_sip.



On Fri, 23 Oct 2020 at 00:13, George Joseph
mailto:gjos...@digium.com>> wrote:



On Wed, Oct 21, 2020 at 9:16 PM David Cunningham
mailto:dcunning...@voisonics.com>> wrote:

Hello,

We have an Asterisk server with two public IP
addresses, let's say 1.1.1.1 and 2.2.2.2. Normally
calls come in to 1.1.1.1 and are bridged with a
call dialled from Asterisk to an external
destination. The external destination sees the SIP
packet as coming from 1.1.1.1 and the media
address in the SDP is 1.1.1.1, which is great.

However if we receive a call in to 2.2.2.2 then
the call dialled from Asterisk to an external
destination still comes from 1.1.1.1, whereas we
want it to come from 2.2.2.2. The source of any
dialled call (the IP packet and the SDP media
address) should be the same as the address the
related inbound call was received to.

For example:
INVITE received to 1.1.1.1:5060
<http://1.1.1.1:5060> -> Asterisk dials
destinat...@termination.com
<mailto:destinat...@termination.com> -> INVITE
sent from 1.1.1.1:5060 <http://1.1.1.1:5060> to
termination.com <http://termination.com>
INVITE received to 2.2.2.2:5060
<http://2.2.2.2:5060> -> Asterisk dials
destinat...@pstn.com <mailto:destinat...@pstn.com>
-> INVITE sent from 2.2.2.2:5060
<http://2.2.2.2:5060> to pstn.com <http://pstn.com>

Does anyone know how this can be achieved?


If termination.com <http://termination.com> is only on
1.1.1.1 and pstn.com <http://pstn.com> is only on
2.2.2.2, create 2 transports, one specifically bound
to 1.1.1.1, transport-1.1.1.1 for instance, and
another to 2.2.2.2 <http://2.2.2.2>:
transport-2.2.2.2.  The names aren't important as long
as you can tell the difference.  Then explicitly
configure endpoint termination.com
<http://termination.com>'s "transport" parameter to
"transport-1.1.1.1" and pstn.com <http://pstn.com>'s
"transport" parameter to "transport-2.2.2.2".   In
your dialplan, you can see which endpoint the call
came in on, and route it out the same endpoint.

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[Jeff LaCoursiere]

I didn't want to post this because its kind of ugly, but we *did* 
actually do it a number of years ago to get around this issue with chan_sip.


Our original architecture was based on LXC, and we had large servers 
running hundreds of containers, each running asterisk.  The "host" ran 
asterisk too, as the gateway for all the container instances.


We once used two of those containers to run asterisk on specific host 
interfaces (one instance bridged to one nic, the other to the other).  
The host asterisk would route calls out one container or the other, with 
the effect you are looking for...


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.


On 10/29/20 7:42 PM, David Cunningham wrote:

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a 
specific IP address for its end of the communication for a specific 
device? Something like:


[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There is the 
"extenip" setting, but it's really designed for NAT, and can only 
appear in the [general] section.


Any suggestions would be greatly appreciated.


On Sat, 24 Oct 2020 at 09:43, David Cunningham 
mailto:dcunning...@voisonics.com>> wrote:


OK, thank you George.


On Sat, 24 Oct 2020 at 03:16, George Joseph mailto:gjos...@digium.com>> wrote:



On Thu, Oct 22, 2020 at 4:13 PM David Cunningham
mailto:dcunning...@voisonics.com>>
wrote:

Hi George,

Thank you for the response. I'm a little unclear on what
you mean by a transport. We're using chan_sip, not pjsip.

Do you mean a device in sip.conf, using bindaddr to set
the address to bind for that device? We've only used
bindaddr in the [general] section before, but if it will
work in a device that could be the answer.


Sorry.  I just assume chan_pjsip these days.  Not sure how
you'd do it for chan_sip.



On Fri, 23 Oct 2020 at 00:13, George Joseph
mailto:gjos...@digium.com>> wrote:



On Wed, Oct 21, 2020 at 9:16 PM David Cunningham
mailto:dcunning...@voisonics.com>> wrote:

Hello,

We have an Asterisk server with two public IP
addresses, let's say 1.1.1.1 and 2.2.2.2. Normally
calls come in to 1.1.1.1 and are bridged with a
call dialled from Asterisk to an external
destination. The external destination sees the SIP
packet as coming from 1.1.1.1 and the media
address in the SDP is 1.1.1.1, which is great.

However if we receive a call in to 2.2.2.2 then
the call dialled from Asterisk to an external
destination still comes from 1.1.1.1, whereas we
want it to come from 2.2.2.2. The source of any
dialled call (the IP packet and the SDP media
address) should be the same as the address the
related inbound call was received to.

For example:
INVITE received to 1.1.1.1:5060
<http://1.1.1.1:5060> -> Asterisk dials
destinat...@termination.com
<mailto:destinat...@termination.com> -> INVITE
sent from 1.1.1.1:5060 <http://1.1.1.1:5060> to
termination.com <http://termination.com>
INVITE received to 2.2.2.2:5060
<http://2.2.2.2:5060> -> Asterisk dials
destinat...@pstn.com <mailto:destinat...@pstn.com>
-> INVITE sent from 2.2.2.2:5060
<http://2.2.2.2:5060> to pstn.com <http://pstn.com>

Does anyone know how this can be achieved?


If termination.com <http://termination.com> is only on
1.1.1.1 and pstn.com <http://pstn.com> is only on
2.2.2.2, create 2 transports, one specifically bound
to 1.1.1.1, transport-1.1.1.1 for instance, and
another to 2.2.2.2 <http://2.2.2.2>:
transport-2.2.2.2.  The names aren't important as long
as you can tell the difference.  Then explicitly
configure endpoint termination.com
<http://termination.com>'s "transport" parameter to
"transport-1.1.1.1" and pstn.com <http://pstn.com>'s
"transport" parameter to "transport-2.2.2.2".   In
your dialplan, you can see which endpoint the call
came in on, and route it out the same endpoint.

Re: [asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

[Jeff LaCoursiere]

I didn't want to post this because its kind of ugly, but we *did* 
actually do it a number of years ago to get around this issue with chan_sip.


Our original architecture was based on LXC, and we had large servers 
running hundreds of containers, each running asterisk.  The "host" ran 
asterisk too, as the gateway for all the container instances.


We once used two of those containers to run asterisk on specific host 
interfaces (one instance bridged to one nic, the other to the other).  
The host asterisk would route calls out one container or the other, with 
the effect you are looking for...


Cheers,

Jeff LaCoursiere
StratusTalk, Inc.


On 10/29/20 9:05 PM, David Cunningham wrote:

Hi Dovid,

We can change the SDP in Kamailio, but Asterisk will still send its 
RTP from its default address. The remote end is strict about accepting 
RTP from the specified source and won't accept it. Have you any 
suggestions to solve that problem?


Thank you.


On Fri, 30 Oct 2020 at 14:49, Dovid Bender <mailto:do...@telecurve.com>> wrote:


Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1
you pass it along as is. Where you want 2.2.2.2 change the sdp in
opensips/kamailio

On Thu, Oct 29, 2020 at 20:44 David Cunningham
mailto:dcunning...@voisonics.com>> wrote:

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a
specific IP address for its end of the communication for a
specific device? Something like:

[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There
is the "extenip" setting, but it's really designed for NAT,
and can only appear in the [general] section.

Any suggestions would be greatly appreciated.



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Re: [asterisk-users] Stir Shaken

[Jeff LaCoursiere]

On 7/13/20 2:32 PM, Saint Michael wrote:


There is a big confusion here about Stir Shaken. It is NOT a
provider issue. Un fact, all providers are whasing their hands and
modifying their swihtches to pass-through the Signature. They
cannot sign the call because then the become the responsible party
for the call before the FCC, and liable for any illegal call.

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


We supply dialtone to a large number of businesses.  We buy DIDs from 
carriers and resell them.  It *may* be up to us to get our direct 
customers' calls signed, but at the moment we are in talks with our DID 
providers to do so on our behalf.  In the next year I have no doubt if 
there are niches to be filled in providing CA or outright 
signing-as-a-service, businesses will be jumping out of the woodwork to 
provide it.  I'm not panicking yet.


  I am the only game in the world for Stir-Shaken and Asterisk. I know 
it sounds arrogant but it is literally true. If you need to sign your 
calls to get through, with Asterisk, you need to connect to my 
service. I am an approved Service Provider from the FCC. If you keep 
thinking this is not happening, it is, and your business will 
disappear overnight.


Its not just arrogant, its silly, and you have a serious branding 
problem.  If you really have "The Answer" you should work on getting 
yourself a domain name at least.  Cease the panic-inducing posts and 
come up with some reasonable fodder you could link to in your signature 
or something (like when you help with some thread), so you would at 
least contribute to the list at the same time.


Some of us may actually be interested in what you have to offer if you 
changed the way you were presenting it.  Who is going to base their 
business on some list guy with a gmail address?


--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell

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Re: [asterisk-users] Voice broken during calls (again...)

[Jeff LaCoursiere]

Hi Luca,

On 6/23/20 8:02 AM, Luca Bertoncello wrote:


I have problem calling someone outside my networks and I have problem 
if the peers are in different networks...


I may have missed this originally - are you saying you have trouble when 
internal phones call each other, if they are on different VLAN's?  
That's a pretty big deal.


I didn't see my post with the graphs of inter-packet latency make it to 
the list (moderator?), I think the images were too large.  Recall that 
clearly showed half of the packets coming inbound from DT were 
*missing*, which confirms your audio experience.  I don't think that 
fact has been addressed properly - it is the only smoking gun you have 
so far.  If that is also happening inter-VLAN, something is seriously 
wrong on the Pi.


If you can reproduce this can you send me a few more packet traces, from 
each of the VLAN interfaces involved?


Always looking for real-world data to improve our tools :)

Cheers,

--

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>



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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

 ability to keep up with 
everything you are asking it to do, when looked at from the 
*microsecond* perspective.


Still doesn't explain the lack of traffic from DT... I would call them, 
send them the trace you sent me, and this message.


Good luck!

Cheers,

j



On 6/15/20 3:27 PM, Luca Bertoncello wrote:

Am 15.06.2020 um 21:50 schrieb Luca Bertoncello:


What do you mean now? If I can use the full available band or if I can
download exactly 50Mbs?
The answer to the first question is: YES! That's why I use a traffic
shaper... ;)
The answer to the second question is: NO. I made a speedtest right now
and I get only ~18Mbps download.

And some other information, too.

I checked the xDSL-statistics of my DSL-Modem (which use the BananaPI to
establish the PPPoE connection):

adsl: ADSL driver and PHY status
Status: Showtime
Last Retrain Reason:2
Last initialization procedure status:   0
Max:Upstream rate = 1709 Kbps, Downstream rate = 19888 Kbps
Bearer: 0, Upstream rate = 1626 Kbps, Downstream rate = 20113 Kbps
Bearer: 1, Upstream rate = 0 Kbps, Downstream rate = 0 Kbps

So it seems, that my connection is about the half of the theorical one...

I think, I must call Deutsche Telekom, but since I'll change my contract
at 18.06., I'll wait some days. Then I'll have a "business" contract,
and I hope I don't must speak with someone that can just say "you have
to reboot your Fritzbox. What? You don't have a Fritzbox? That's not
possible. Please check your Fritbox, I can't reach it"... ;)

Bye
Luca Bertoncello
(lucab...@lucabert.de)



--

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>



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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

On 6/16/20 1:18 AM, Luca Bertoncello wrote:

Am 15.06.2020 23:15, schrieb Jeff LaCoursiere:

Hi again,

just a question, to be sure...


sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &


eth0 is my DSL interface and eth1 my phone interface?



Sure, that's fine.  We will figure out which one is north/south in the 
analysis.



Try to limit the traffic to just your phone call tests (to reduce the
size of the capture files).  Make all your tests, then:


Well, assuming eth0 is the DSL interface and eth1 the phone interface, 
I can so that:


tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de &
tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of 
my phone) &


is it correct?



Perfect.

Cheers,

j



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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

On 6/15/20 2:19 PM, Luca Bertoncello wrote:

Am 15.06.2020 um 20:15 schrieb Jeff LaCoursiere:

Hi Jeff,


We are working on a product to analyze pcap files of VoIP calls.  So far
it does a reasonable job of analyzing the frequency distribution of
packets in both directions, pointing out which direction packet loss /
bad jitter occurs.  If you can trap the traffic on the outside and the
inside of your Banana Pi and send me the pcap files, I would be happy to
run it through our analyzer as further information for you.  If it shows
DTK isn't sending packets when it should, that will be obvious, and you
can send to them as solid evidence of their guilt :)

Thank you for your offer.
Could you say me which options I should pass to tcpdump to get all
information you need?

Yes, sure, please use (replace with correct interface names):

   sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
   sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &


Try to limit the traffic to just your phone call tests (to reduce the 
size of the capture files).  Make all your tests, then:


   sudo killall tcpdump
   tar cvzf /tmp/tests.tgz /tmp/test?.pcap


Send /tmp/tests.tgz to me by email, or leave somewhere I can download.  
I'll run the analysis tonight and send the results to the list.


Cheers,

--

    *Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>



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Re: [asterisk-users] Voice "broken" during calls

[Jeff LaCoursiere]

Hi,

We are working on a product to analyze pcap files of VoIP calls. So far 
it does a reasonable job of analyzing the frequency distribution of 
packets in both directions, pointing out which direction packet loss / 
bad jitter occurs.  If you can trap the traffic on the outside and the 
inside of your Banana Pi and send me the pcap files, I would be happy to 
run it through our analyzer as further information for you.  If it shows 
DTK isn't sending packets when it should, that will be obvious, and you 
can send to them as solid evidence of their guilt :)


Beyond that, are you certain you aren't taxing the Banana Pi?  It really 
*should* be able to handle a single call while handling your LAN's 
routing/firewall tasks, but you are probably skating the edge.  The 
results of the above might point out that the Pi isn't *sending* packets 
it should be, or sending them way late, in which case the issue is 
actually your hardware.


Cheers,

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 6/15/20 11:55 AM, Luca Bertoncello wrote:

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello:

Hi

So, I got a phone (Elmeg IP290) from a collegue and tested it...


What I'll do tomorrow with a test phone is:

1) connecting it to my Asterisk and try to make a call
2) connecting it directly to the servers of Deutsche Telekom (using my
network) and try to make a call

Absolutly *no changes* on the behaviour compared with my Thomsons...

I try to summarize:

1) Phones are not the problem, since 3 phones of 2 different
companies/model have the same issue.
2) Asterisk seems not to be the problem, too, since I have the same
behaviour if I connect to phone directly to the server of Deutsche Telekom.
3) Traffic shaping seems not to be the problem, too, since I tried to
deactivate it.
4) The problem happens *only* on active call, not by voicemail.
4a) To test it I read a text and my partner just listen it, and then he
read a text and I listen it. *No* simulaneously speak!
5) A *single call* (since I couldn't reproduce it anymore), made using
my Android phone as SIP-client connected to my Asterisk, had not the
problem. Any other try to call someone using my mobile phone via SIP had
the problem.

I could *not* test connecting to the server of Deutsche Telekom using
the Internet connection of someone other, since Telekom bounds my
VoIP-login to my IP.

I really think, the problem should be by Deutsche Telekom...

What is your opinion? Do you see some other tests I should try?

Thanks a lot
Luca Bertoncello
(lucab...@lucabert.de)

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Re: [asterisk-users] CDR mysql: timeout when remote database unavailable

[Jeff LaCoursiere]

Isn't the MySQL stuff deprecated in favor of odbc?  You may be barking up
the wrong tree if you plan to make source changes.

j

On Sun, Jun 7, 2020, 1:55 AM Fourhundred Thecat <400the...@gmx.ch> wrote:

>  > On 2020-06-06 10:38, Antony Stone wrote:
> > On Saturday 06 June 2020 at 09:18:11, Fourhundred Thecat wrote:
> >
> >> In a situation when I start asterisk, and the remote database is
> >> unreachable, asterisk waits for several minutes before it actually
> >> starts (before it loads sip module, etc).
> >>
> >> And when database is unreachable during operation, when call happens,
> >> sometimes the call is connected, other times it waits for mysql and call
> >> times out.
> >
> >> or what would be the best solution in my case ?
> >
> > I would install a local copy of MySQL (to the same machine as Asterisk)
> so
> > that it is definitely available, tell Asterisk to write to that, and
> then set
> > up replication to the remote MySQL instance which is sometimes
> unavailable.
>
> That is an interesting suggestion, which I'll consider.
>
> But I would still like to know where the Aterisk mysql timeout duration
> comes from, and whether it can be configured.
>
> thanks,
>
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Re: [asterisk-users] CLI color prompt

[Jeff LaCoursiere]

I work from a similar setup.  I ssh'ed to my personal PBX from an xterm 
window on an Ubuntu 16 workstation, your prompt seems to work:



*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 5/31/20 8:59 AM, Antony Stone wrote:

On Sunday 31 May 2020 at 15:44:46, Fourhundred Thecat wrote:


Hello,

how can I change the color of the asterisk prompt to red ?

I read in the wiki that I can use %Cn[;n]

https://wiki.asterisk.org/wiki/display/AST/Asterisk+CLI+Configuration

"The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable that
you set from the Unix shell before starting Asterisk."


I currently have this in my environment:

export ASTERISK_PROMPT="[%H]: "

which changes the prompt to hostname

Ho can I make this prompt red ?

"%Cn[;n] - Change terminal foreground (and optional background) color to
specified A full list of colors may be found in include/asterisk/term.h"

So, try:

export ASTERISK_PROMPT="%C31[%H]: "

(I got 31 from reading the term.h file.)


Regards,


Antony.

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Re: [asterisk-users] CLI color prompt

[Jeff LaCoursiere]

Hi,

I had posted this a few hours ago, but got caught in moderation for 
size.  I trimmed down the pic and attached.


I am on an Ubuntu 16 workstation, in an Ubuntu terminal window, ssh'ed 
to the PBX (amazon instance).  You can see my term type matches yours.


I really don't know why yours doesn't work.  Perhaps you can tell us 
what your terminal emulator is, what you are running it on, etc.  
Whatever it is, it isn't properly interpreting the escape codes for 
xterm-256color.  You could possibly try some different terminal types, 
but this is an odd situation if the remote shell (is it remote?) can't 
determine your termtype.  This is pretty ancient code.


j


*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 5/31/20 11:42 AM, Fourhundred Thecat wrote:

On 2020-05-31 18:39, Ira wrote:

I typed this at the terminal prompt:  export ASTERISK_PROMPT="%C31[%H]: "

Typing at the same place:  echo $TERM  returns xterm

And now I have colored prompts at the Asterisk command line, so I can
assure you it can work. Kind of cool, 14 years using Asterisk and
because of your question, I now have colored prompts.

Do I have to do something to make sure that ASTERISK_PROMPT lives
through a reboot?

I would add the export to .bashrc/.zshrc or whatever shell you are using

why does it not work for me?
My terminal clearly supports 256 colors.

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Re: [asterisk-users] CLI color prompt

[Jeff LaCoursiere]

I'm pretty sure that means your are using a non-color capable terminal, 
or your termtype variable is incorrect.  What are you using for a 
terminal emulator?


*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 5/31/20 9:03 AM, Fourhundred Thecat wrote:

> On 2020-05-31 15:59, Antony Stone wrote:

On Sunday 31 May 2020 at 15:44:46, Fourhundred Thecat wrote:

"%Cn[;n] - Change terminal foreground (and optional background) color to
specified A full list of colors may be found in include/asterisk/term.h"

So, try:

export ASTERISK_PROMPT="%C31[%H]: "

(I got 31 from reading the term.h file.)


thanks, but that does not work for me. When I use your example:

export ASTERISK_PROMPT="%C31[%H]: "

I get this prompt (voip is my hostname):

[1;31m[voip]: [1;0m

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Re: [asterisk-users] Stir-Shaken for asterisk

[Jeff LaCoursiere]

A few weeks... like in a year and a few weeks:

https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/

Some interesting bits in there as well, like:

"These rules do not apply to providers that lack control of the network 
infrastructure necessary to implement STIR/SHAKEN."


See also:

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


    *Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 5/27/20 10:51 PM, Saint Michael wrote:
In a few weeks, no SIP call is going to terminate unless they 
are signed properly, as mandated by law.  We are in the business of 
Stir-Shaken, signing calls, as an FCC-approved provider. A big 
differentiator between our service and the rest: we are the only ones 
who don't need to receive the calls in our servers to sign them. We do 
this over a MySQL call, easily connectable to Asterisk via res_odbc, 
so you never have to send us your calls. This is a sample of how we do 
this so you may test now:
mysql -u anonymous -h 208.73.232.47 -e "call 
strshk.stir_shaken_signature('7274433019','1957408')".
If your caller-ID is a valid US number and not a wireless number (that 
is a NO-NO for the FCC), we sign the call as 'C', if you use your own 
DIDs, something we can verify as legit, then we sign as 'B', and if 
you use our DID as caller ID, we sign as 'A', full attestation.
Please email to venefax at g mail if you have any questions. Do not 
think you can do business as usual. The wild west of VOIP is coming to 
an end. But we can keep you in business if you follow the rules.


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Re: [asterisk-users] Stir-Shaken for asterisk

[Jeff LaCoursiere]

In a few weeks?  FIrst I have heard of this, and your legitimacy is 
strained by a gmail address.


*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 5/27/20 10:51 PM, Saint Michael wrote:
In a few weeks, no SIP call is going to terminate unless they 
are signed properly, as mandated by law.  We are in the business of 
Stir-Shaken, signing calls, as an FCC-approved provider. A big 
differentiator between our service and the rest: we are the only ones 
who don't need to receive the calls in our servers to sign them. We do 
this over a MySQL call, easily connectable to Asterisk via res_odbc, 
so you never have to send us your calls. This is a sample of how we do 
this so you may test now:
mysql -u anonymous -h 208.73.232.47 -e "call 
strshk.stir_shaken_signature('7274433019','1957408')".
If your caller-ID is a valid US number and not a wireless number (that 
is a NO-NO for the FCC), we sign the call as 'C', if you use your own 
DIDs, something we can verify as legit, then we sign as 'B', and if 
you use our DID as caller ID, we sign as 'A', full attestation.
Please email to venefax at g mail if you have any questions. Do not 
think you can do business as usual. The wild west of VOIP is coming to 
an end. But we can keep you in business if you follow the rules.


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Re: [asterisk-users] Block Spam Calls

[Jeff LaCoursiere]

Mind posting your dialplan code?  I was thinking the same thing - very 
much like an old spam control program I used to use whose name now 
escapes me.  First time senders would have to respond to an auto-reply, 
then were added to a whitelist.


This would be a great FreeBSD module...

Cheers,

*Jeff LaCoursiere*
STRATUSTALK, INC. / CTO

Phone:  *+1 703.496.4990 x108*
Mobile: *+1 815.546.6599*
Email:  *j...@stratustalk.com* <mailto:j...@stratustalk.com>
Website:*https://www.stratustalk.com*
Address:*One Freedom Square
13th Floor
Reston, VA 20190*

<https://www.facebook.com/jeff.lacoursiere> 
<https://linkedin.com/in/jeff-lacoursiere-884361> 
<https://www.twitter.com/stratustalk>


On 12/13/19 10:48 AM, Julian Beach wrote:

Hello Doug,

Friday, December 13, 2019, 11:03:37 AM, you wrote:


This is exactly what I do - “press 1 for a human”
Works great

I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call directly on without
the IVR prompts.

I do something similar for calls without caller ID, but I was still
getting robocalls with spoofed caller ID. I have now changed the dialplan
slightly so that the first time people call they are asked to dial 1.
After the first call, they are added to a known caller list and get
straight through, and any robocalls at that point are blacklisted
manually. I have found that most robocallers spoof the Caller ID so
rarely call from the same number twice. It means that legitimate
callers who cannot dial 1 just have to dial again to get through to
the phones - there is a recorded message telling them to dial 1 or
call back. I haven't had a robocall since!

The hardest thing about this was extracting all the numbers of
previous callers from the CDR and adding it to the Previous_Callers
AstDB for the lookup. I didn't want to make existing callers go through
the initial learning process.

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[asterisk-users] Delayed RTP start

[Jeff LaCoursiere]

Hi,

I am debugging an issue that unfortunately involves two NAT instances - 
the firewall at our customer site, and the firewall in front of their 
Amazon instance.


I have an HTEK phone at the customer site registering to the public 
address of the Amazon instance running asterisk (and FreePBX).  This 
seems to work fine, and it can call local services (like fpbx *65 to 
read back the extension) with no problems.


If it tries to make an outbound outside call, the remote phone (my cell 
for example) rings, I answer it, but there is no audio in either 
direction for nearly exactly 16 seconds, every time.  Then audio starts 
in both directions without issue.


I did a packet trace on the phone itself and see 16 seconds of outbound 
RTP with no inbound, then suddenly RTP in both directions until the call 
ends.


I did a packet trace on the asterisk side and see the call setup, then 
sixteen seconds of nothing (??), then RTP starts in both directions.


In the asterisk console I see this bit of interestingness:

[2019-07-24 13:21:02] DEBUG[1890]: chan_sip.c:29923 
__start_session_timer: Session timer started: 78 - 
710779684e62266a77b047b31e4

261da@10.0.116.239:60060 1768000ms
    -- SIP/ast01-024b answered SIP/7222-024a

[.snip.]

[2019-07-24 13:21:02] DEBUG[17928][C-01f1]: bridge_native_rtp.c:660 
native_rtp_bridge_compatible_check: Bridge '3bfbf253-d34f-
45e2-abc3-75e590d81739' can not use native RTP bridge as channel 
'SIP/ast01-024b' has DTMF hooks


[.snip.]

[2019-07-24 13:21:18] DEBUG[18003][C-01f1]: res_rtp_asterisk.c:4179 
ast_rtp_write: Ooh, format changed from none to ulaw
[2019-07-24 13:21:18] DEBUG[18003][C-01f1]: res_rtp_asterisk.c:4019 
rtp_raw_write: Starting RTCP transmission on RTP instan

ce '0x7fe17426e7c8'


So my main question is, what would cause a sixteen second delay before 
the codec could be decided?


This is Asterisk 13.25.0 on the customer Amazon instance... the "ast01" 
peer is ours also - one of our external gateways, also running 13.25.0.


Thanks,


--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell


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Re: [asterisk-users] Hacking

[Jeff LaCoursiere]

Our provisioning servers listen on a high numbered port.  We generally 
don't have any issues with scanning...


Cheers,

j

On 6/18/19 7:18 AM, John Runyon wrote:
Just to jump in on this, this just started happening to our system a 
couple days ago. (To the tune of 3GB of webserver access logs yesterday)
Our server gives them a 403 for /yealink/ (and a 404 for everything 
else) - given that they're still trying to bruteforce it, it looks 
like I'm gonna be changing it to give them a 404.
Looks like someone's making a big effort to find provisioning files 
though.


On Mon, Jun 17, 2019, 13:35 John Kiniston > wrote:




On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner mailto:j...@xaccel.net>> wrote:

Anyone know how someone can hack an asterisk box and register
with every single account on the box.

This box only has 3 accounts, with very complex passwords.
Have VoIP blacklist setup and fail2ban…


I've seen this happen when web-based provisioning is used, I have
seen attempts to download configuration files off of my
provisioning server increase in frequency over the last two years.

The 'Hacker' will do a get on /polycom /cisco /yealink /aastra
/mitel etc, If they get a valid response they will start
enumerating mac addresses

/polycom/0004F2018101.cfg
/polycom/0004F2018102.cfg
...
/polycom/0004F2018109.cfg

Then they will use any credentials gained in the download attack
to place calls, registering as needed.

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Re: [asterisk-users] DUNDI with minimal features

[Jeff LaCoursiere]

You could use IMAP storage for your voicemail to solve this I think.  
Have both PBXes use the same storage.


I don't know if this works for you or not, but you might consider a 
single PBX and just combine the two offices under one installation.  We 
do this for a lot of our customers (actually we host their PBX, but all 
the offices' phones connect to it).  If both offices have good 
connectivity, and especially if you have a QoS enabled VPN between them, 
this could work well.


Cheers,

j


On 03/27/2019 12:43 PM, Janet wrote:

Great document thank you!  I will have to experiment with this on a couple of 
test systems.

Something I'm not clear on, if a user receives a voicemail on one of the PBX's, 
does DUNDI handle retrieving the message from the right system?  Or if the user 
tries to retrieve a voicemail on PBX A but the message was left on PBX B, they 
won't hear it.

Janet

-Original Message-
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of JR Richardson
Sent: Wednesday, March 27, 2019 2:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DUNDI with minimal features


I have 2 PBX's, one in each office (say one in New York, one in
Boston).  I have mobile users that can show up at either office and
connect their soft phones.



Is there a very simple DUNDI config available which describes how to
set this up?

Also, can I have the same outbound trunks setup in each office, so
that calls don't have to route across the NY-BOS connection to get out
to the PSTN?



Thanks, Janet

Not sure how relevant on newer versions, but yes, pretty easy to setup.

http://blog.sinologic.net/wp-content/uploads/2007/08/DUNDi_So_Easy.pdf

Good luck!.

JR
--
JR Richardson
Engineering for the Masses
Chasing the Azeotrope

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--
Jeff LaCoursiere
StratusTalk, Inc.
703 496 4990 x108
815 546 6599 cell
j...@stratustalk.com


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Re: [asterisk-users] Digium G100

[Jeff LaCoursiere]

Definitely getting the caller id info - see below.  Its just ending up 
in the wrong field.  The caller's number is ending up in the "name" 
field, and the "number" field is getting our G100's SIP peer name.


Its not clear if it is being offered that way to the asterisk server 
that is accepting the call from the G100, or if the asterisk server is 
mangling it before sending it on to the customer...  I can do some 
dialplan foo I suppose to answer this question, but was really hoping 
someone would say "oh I had that problem..." :)


j

On 2/15/19 12:45 PM, Nick Olsen wrote:
You might confirm you're getting CallerID from the PRI in the call 
setup. You can do a debug capture session on the G100 and get this info.


If you need CallerID preserved from the PRI (Like the served PBX sends 
multiple calling numbers based on end user station) then you'll likely 
need to fix it on whatever the G100 is serving with said PRI.


If it's all one number anyways, You can just blanket overwrite it from 
the G100 dialplan (I think it was in outbound routes). Or ultimately, 
In the asterisk instance during receive before shooting it upstream.



*Nick Olsen*
Network Engineer
Office: 321-408-5000
Mobile: 321-794-0763

--------
*From*: Jeff LaCoursiere 
*Sent*: 2/15/19 1:12 AM
*To*: Asterisk Users Mailing List - Non-Commercial Discussion 


*Subject*: [asterisk-users] Digium G100
Hi,

We recently dumped a Xorcom box that was no end of trouble and replaced
with a Digium G100.  PRI came right up, and we have been using it fairly
flawlessly for several months now, with one caveat.  Calls that arrive
from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then
routed by the dialplan to various other gateways or upstream providers.
When the call finally lands on a phone somewhere, the caller ID
information has become corrupted, though in a predictable way.

The CID number is replaced with the SIP trunk name of our G100 gateway.

The CID name is replaced by the callers phone number.

This is problematic for a number of reasons - we have lost the caller ID
name, if provided, completely.  There is a lot of confusion from our
customers asking "what does riisegw mean?!", and if they try to return a
missed phone call or recall something from their history, their phones
(Yealink models almost exclusively) try to dial to "riisegw" since that
was actually in the number field.

I haven't tried to dig into this on our asterisk instance yet, was
hoping this is something silly someone could direct us to, or perhaps
someone from Digium can pitch in.  I suppose I should have some kind of
support with the G100... have never tried to actually call Digium before.

Cheers,

Jeff LaCoursiere


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[asterisk-users] Digium G100

[Jeff LaCoursiere]

Hi,

We recently dumped a Xorcom box that was no end of trouble and replaced 
with a Digium G100.  PRI came right up, and we have been using it fairly 
flawlessly for several months now, with one caveat.  Calls that arrive 
from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then 
routed by the dialplan to various other gateways or upstream providers.  
When the call finally lands on a phone somewhere, the caller ID 
information has become corrupted, though in a predictable way.


The CID number is replaced with the SIP trunk name of our G100 gateway.

The CID name is replaced by the callers phone number.

This is problematic for a number of reasons - we have lost the caller ID 
name, if provided, completely.  There is a lot of confusion from our 
customers asking "what does riisegw mean?!", and if they try to return a 
missed phone call or recall something from their history, their phones 
(Yealink models almost exclusively) try to dial to "riisegw" since that 
was actually in the number field.


I haven't tried to dig into this on our asterisk instance yet, was 
hoping this is something silly someone could direct us to, or perhaps 
someone from Digium can pitch in.  I suppose I should have some kind of 
support with the G100... have never tried to actually call Digium before.


Cheers,

Jeff LaCoursiere


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[asterisk-users] Digium T1 gateway caller ID issues

[Jeff LaCoursiere]

Hi,

We recently dumped a Xorcom box that was no end of trouble and replaced 
with a Digium G100.  PRI came right up, and we have been using it fairly 
flawlessly for several months now, with one caveat.  Calls that arrive 
from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then 
routed by the dialplan to various other gateways or upstream providers.  
When the call finally lands on a phone somewhere, the caller ID 
information has become corrupted, though in a predictable way.


The CID number is replaced with the SIP trunk name of our G100 gateway.

The CID name is replaced by the callers phone number.

This is problematic for a number of reasons - we have lost the caller ID 
name, if provided, completely.  There is a lot of confusion from our 
customers asking "what does riisegw mean?!", and if they try to return a 
missed phone call or recall something from their history, their phones 
(Yealink models almost exclusively) try to dial to "riisegw" since that 
was actually in the number field.


I haven't tried to dig into this on our asterisk instance yet, was 
hoping this is something silly someone could direct us to, or perhaps 
someone from Digium can pitch in.  I suppose I should have some kind of 
support with the G100... have never tried to actually call Digium before.


Cheers,

Jeff LaCoursiere


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[asterisk-users] Xorcom PRI

[Jeff LaCoursiere]

I've been struggling for a few weeks now with the local telco trying to 
bring up a trunk that has been down for a year (hurricanes in the 
caribbean).  Box is a Dell R710, 16G RAM, Ubuntu 14.04.5 LTS, Dahdi 
2.10.2-rc1, asterisk 13.23.1.  Xorcom Astribank w/ one T1/E1/PRI module, 
plugged into a USB 2.0 port on the Dell.  All of this was working 
*before* the storms last year with the same hardware/versions.


Dahdi sees the astribank and loads firmware without issue:

   root@astbeach:~# dmesg | grep -i dahdi
   [661368.877090] dahdi: Version: 2.10.2-rc1
   [661368.880450] dahdi: Telephony Interface Registered on major 196
   [661368.963988] dahdi_transcode: Loaded.
   [661368.982746] INFO-xpp: FEATURE: with sync_tick() from DAHDI
   [661369.233471] INFO-xpd_pri: FEATURE: WITHOUT DAHDI_AUDIO_NOTIFY
   [661370.256053] dahdi_devices astribanks:xbus-00: local span 1 is
   already assigned span 1
   [661370.270028] dahdi_echocan_mg2: Registered echo canceler 'MG2'

   root@astbeach:~# lsusb
   Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
   Bus 006 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
   Bus 005 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
   Bus 001 Device 003: ID 0424:2514 Standard Microsystems Corp. USB 2.0 Hub
   Bus 001 Device 002: ID e4e4:1162 Xorcom Ltd. Astribank 2 series
   Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
   Bus 004 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub
   Bus 003 Device 001: ID 1d6b:0001 Linux Foundation 1.1 root hub

The dahdi drivers are loaded, and the T1 layer has no alarms... telco 
also reports the line itself is "UP":


root@astbeach:~# service dahdi status
### Span  1: XBUS-00/XPD-00 "Xorcom XPD [usb:X1067719].1: T1" (MASTER) 
ESF/B8ZS ClockSource

  1 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  2 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  3 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  4 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  5 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  6 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  7 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  8 T1 Clear   (In use) (EC: MG2 - INACTIVE)
  9 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 10 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 11 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 12 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 13 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 14 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 15 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 16 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 17 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 18 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 19 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 20 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 21 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 22 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 23 T1 Clear   (In use) (EC: MG2 - INACTIVE)
 24 T1 Hardware-assisted HDLC  (In use)

asterisk chan_dahdi shows the T1 up with no alarms:

astbeach*CLI> dahdi show status
Description  Alarms  IRQ bpviol CRC    Fra 
Codi Options  LBO
Xorcom XPD [usb:X1067719].1: T1  OK  0 0  0  ESF 
B8ZS  0 db (CSU)/0-133 feet (DSX-1)


but the PRI is down:

astbeach*CLI> pri show spans
PRI span 1/0: Down, Active

I'm not really sure where to take it from here, and the telco has even 
less of a clue.  They brought out some gear that they hooked up to our 
cabling for the T1 and pretty quickly established a PRI, then placed and 
received test calls over it.  At that point they washed their hands of 
it, and logged as a "CPE issue"!


Could it be that the storms damaged the Xorcom unit in such a way that 
the T1 can be up without alarms but the PRI signaling is broken?  Seems 
unlikely.


I have included a few relevant config files below.  Note that the 
cabling wasn't in place when we ran dahdi_genconf, which is why it shows 
red alarm.  There is no red alarm now.


   /etc/dahdi/system.conf:

   # Autogenerated by /usr/sbin/dahdi_genconf on Fri Oct 12 11:34:27 2018
   # If you edit this file and execute /usr/sbin/dahdi_genconf again,
   # your manual changes will be LOST.
   # Dahdi Configuration File
   #
   # This file is parsed by the Dahdi Configurator, dahdi_cfg
   #
   # Span 1: XBUS-00/XPD-00 "Xorcom XPD [usb:X1067719].1: T1" (MASTER) RED
   span=1,1,0,esf,b8zs
   # termtype: te
   bchan=1-23
   #dchan=24
   echocanceller=mg2,1-23
   hardhdlc=24

   # Global data

   loadzone    = us
   defaultzone    = us

   --

   root@astbeach:/etc/dahdi# egrep -v '^#' xpp.conf
   pri_protocol    T1

   --

   

Re: [asterisk-users] multi step auth?

[Jeff LaCoursiere]

Thats till doesn't change the SIP header.  Basically they want to send a 
RE INVITE and authenticate my DID number.  But my DID number does not 
have a peer or user entry in sip.conf.  Perhaps I am answering my own 
question, but is that the only way this is going to work?


Thanks,

j

On 05/08/2018 02:54 PM, Khalil Khamlichi wrote:

try adding a + sign for the number

same => n,Set(CALLERID(all)=17864089672 <+17864089672>)




On Tue, May 8, 2018, 8:51 PM Jeff LaCoursiere <j...@stratustalk.com 
<mailto:j...@stratustalk.com>> wrote:



I *am* doing that, as I assumed it would be required just for the
911 mapping we have provided, but that doesn't change the SIP header.

Cheers,

j

On 05/08/2018 02:41 PM, Khalil Khamlichi wrote:

try setting the callerid with

same => n,Set(CALLERID(all)=17864089672 <17864089672>)

ofcourse for each customer you will need to provide his own did.


    On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere
<j...@stratustalk.com <mailto:j...@stratustalk.com>> wrote:

Hi,

We have been using Voxbone for some time for origination, and
they now offer E911 services.  We are trying to set this up
and having trouble meeting their authentication requirements.

I setup a peer as I normally would, with user/pass as they
supplied ("lacoursj", "pass"), but my calls are rejected. 
Their support is asking that I follow this auth mechanism:

1st step - You send an INVITE message.
2nd step - We respond with a 407.
3rd step - You send a RE INVITE message including your
credentials.

 The tricky bit seems to be that they want the original
INVITE to look like:

From: <sip:*17864089672*@X.X.X.X:60060>;tag=as00771983.
To: <sip:7...@voxout.voxbone.com>
<mailto:sip:7...@voxout.voxbone.com>.
Contact: <sip:*17864089672*@X.X.X.X:60060>.

The "1786..." above is meant to be the DID number that is
placing the 911 call. Our DID numbers don't have peer or user
entries in sip.conf. My peer isn't sending that, though, it
is sending:

From: <sip:*lacoursj*@X.X.X.X:60060>;tag=as00771983.
To: <sip:7...@voxout.voxbone.com>
<mailto:sip:7...@voxout.voxbone.com>.
Contact: <sip:*lacoursj*@X.X.X.X:60060>.

They claim that 'lacoursj' shouldn't be sent until step 3.

I have never been asked to authenticate this way... can
asterisk chan_sip do it?

Cheers,

j
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Re: [asterisk-users] multi step auth?

[Jeff LaCoursiere]

I *am* doing that, as I assumed it would be required just for the 911 
mapping we have provided, but that doesn't change the SIP header.


Cheers,

j

On 05/08/2018 02:41 PM, Khalil Khamlichi wrote:

try setting the callerid with

same => n,Set(CALLERID(all)=17864089672 <17864089672>)

ofcourse for each customer you will need to provide his own did.


On Tue, May 8, 2018, 8:37 PM Jeff LaCoursiere <j...@stratustalk.com 
<mailto:j...@stratustalk.com>> wrote:


Hi,

We have been using Voxbone for some time for origination, and they
now offer E911 services.  We are trying to set this up and having
trouble meeting their authentication requirements.

I setup a peer as I normally would, with user/pass as they
supplied ("lacoursj", "pass"), but my calls are rejected. Their
support is asking that I follow this auth mechanism:

1st step - You send an INVITE message.
2nd step - We respond with a 407.
3rd step - You send a RE INVITE message including your credentials.

 The tricky bit seems to be that they want the original INVITE to
look like:

From: <sip:*17864089672*@X.X.X.X:60060>;tag=as00771983.
To: <sip:7...@voxout.voxbone.com> <mailto:sip:7...@voxout.voxbone.com>.
Contact: <sip:*17864089672*@X.X.X.X:60060>.

The "1786..." above is meant to be the DID number that is placing
the 911 call. Our DID numbers don't have peer or user entries in
sip.conf. My peer isn't sending that, though, it is sending:

From: <sip:*lacoursj*@X.X.X.X:60060>;tag=as00771983.
To: <sip:7...@voxout.voxbone.com> <mailto:sip:7...@voxout.voxbone.com>.
Contact: <sip:*lacoursj*@X.X.X.X:60060>.

They claim that 'lacoursj' shouldn't be sent until step 3.

I have never been asked to authenticate this way... can asterisk
chan_sip do it?

Cheers,

j
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[asterisk-users] multi step auth?

[Jeff LaCoursiere]

Hi,

We have been using Voxbone for some time for origination, and they now 
offer E911 services.  We are trying to set this up and having trouble 
meeting their authentication requirements.


I setup a peer as I normally would, with user/pass as they supplied 
("lacoursj", "pass"), but my calls are rejected.  Their support is 
asking that I follow this auth mechanism:


1st step - You send an INVITE message.
2nd step - We respond with a 407.
3rd step - You send a RE INVITE message including your credentials.

 The tricky bit seems to be that they want the original INVITE to look 
like:


From: ;tag=as00771983.
To: .
Contact: .

The "1786..." above is meant to be the DID number that is placing the 
911 call. Our DID numbers don't have peer or user entries in sip.conf. 
My peer isn't sending that, though, it is sending:


From: ;tag=as00771983.
To: .
Contact: .

They claim that 'lacoursj' shouldn't be sent until step 3.

I have never been asked to authenticate this way... can asterisk 
chan_sip do it?


Cheers,

j
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Re: [asterisk-users] ​ PJSIP and Non Media Proxy

[Jeff LaCoursiere]

On 11/06/2017 12:34 PM, Joshua Colp wrote:

On Mon, Nov 6, 2017, at 02:14 PM, Saint Michael wrote:

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

There is noone to be "told" to do such work. Asterisk is an open source
project that includes not just Digium but also other contributors. It's
all of us working together that helps to improve Asterisk. Like
everything in life we all have our own priorities and responsibilities,
which differ, that drive what we all work on and have an interest in. As
for providing direction for doing the work yourself - this is not
something that anyone has looked into or planned so there's no real
direction to give. You have to pick somewhere to start, dig, and figure
out what needs to be done. If you have specific questions then the
asterisk-dev mailing list would be the best place to discuss such things
since that is where developer talk occurs.

Beyond that, what makes you think forcing asterisk not to proxy RTP 
would be the best way forward for all that use asterisk?  My company 
DEPENDS on it, and direct RTP would simply not function in our 
environment.  Please don't do that.  Having it be configurable is 
obviously a good thing.


j


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[asterisk-users] Openfire and asterisk

[Jeff LaCoursiere]

Anyone have any recent experience with openfire and asterisk 
integration, perhaps with the spark IM client?  About to dive into this 
and would appreciate any advice on gotchas.


Cheers,

j

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Re: [asterisk-users] OT: Want to capture all SIP messages

[Jeff LaCoursiere]

On 05/31/2017 04:13 PM, Steve Edwards wrote:

On Wed, 31 May 2017, Barry Flanagan wrote:


sngrep


Isn't sngrep a great tool? Since discovering it my use of 
tcpdump/wireshark has cratered.


Being able to compare an INVITE that worked with one that didn't (with 
color highlighting) rocks.


On sites where I want an always available packet history I use tcpdump 
with the -C and -W options to manage a ring buffer of X bytes.  Then you 
can use cool tools like sngrep or really anything that operates on pcap 
files at whim.


Cheers,

j

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Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

[Jeff LaCoursiere]

Don't get me wrong - he should absolutely do that.  Just offering an 
explanation for his post.  I read it as "I am trying to install 
certified and it won't compile.  Normal 13 does though.".  He didn't 
mention if the 13 he was trying to compile was certified. Why he would 
want 11 over 13 is anyone's guess.


j

On 04/29/2017 11:15 AM, Jonathan H wrote:

Sure, so why not install the current supported certified 13.13-cert 3
which he confirms builds OK, rather than the about-to-become-EOL 11
version?

http://www.asterisk.org/downloads/asterisk/all-asterisk-versions

On 29 April 2017 at 17:11, Jeff LaCoursiere <j...@jeff.net> wrote:

On 04/29/2017 10:57 AM, Jonathan H wrote:

On 29 April 2017 at 16:47, Tech Support <aster...@voipbusiness.us> wrote:


I’m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16
server. However, when I try to compile it, I’m getting hundreds and hundreds
of errors. Here is a sample of the output.
When I try to build Asterisk 13, I have no problem. Any insight at all
would be greatly appreciated.

I suppose the first and most obvious question would be:

If the current version installs fine, why would you want to install an
old obsolete version which is currently subject to an end of life
warning?

https://community.asterisk.org/t/asterisk-11-eol-6-month-notice/70490

"As many of you know, for the past 6 months Asterisk 11 has been in
security
fix only mode. This means it currently does not receive bug fixes,
but it does receive applicable security fixes and will continue to do
so for the next 6 months."


Probably because he wants to run certified...


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Re: [asterisk-users] configure AudioCodes MP-112 with Asterisk.

[Jeff LaCoursiere]

On 04/29/2017 11:12 AM, the...@sys-concept.com wrote:

I've MP-114 that is working configured and working OK with my Asterisk
but I just obtained MP-112 (2xFXS) and I can register OK with asterisk but I 
can only dial 3-digit extension.

Anything longer than 3-digits is cut off, example I dial extension 1000:

[Apr 29 10:03:30] NOTICE[3817][C-00e9]: chan_sip.c:25902 
handle_request_invite: Call from '54' (10.0.0.115:5060) to extension '100' 
rejected because extension not found in context 'internal'.

My dial plan is working OK as when I register Linsys/Sipura with asterisk 
"context=internal" and I dial any number and dial plan is working OK.

It seems to me Audiocodes MP-112 is trimming anything that is longer than 
3-digits.

Audiocodes has a default dialplan of XXX, which will trim your dialing 
to three digits.  Change it to X+ (or whatever makes sense).


j


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Re: [asterisk-users] Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16

[Jeff LaCoursiere]

On 04/29/2017 10:57 AM, Jonathan H wrote:

On 29 April 2017 at 16:47, Tech Support  wrote:


I’m trying to install certified asterisk 11.6 cert16 on a Ubuntu 16 server. 
However, when I try to compile it, I’m getting hundreds and hundreds of errors. 
Here is a sample of the output.
When I try to build Asterisk 13, I have no problem. Any insight at all would be 
greatly appreciated.

I suppose the first and most obvious question would be:

If the current version installs fine, why would you want to install an
old obsolete version which is currently subject to an end of life
warning?

https://community.asterisk.org/t/asterisk-11-eol-6-month-notice/70490

"As many of you know, for the past 6 months Asterisk 11 has been in security
fix only mode. This means it currently does not receive bug fixes,
but it does receive applicable security fixes and will continue to do
so for the next 6 months."


Probably because he wants to run certified...


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Re: [asterisk-users] AGI Exec Voicemail

[Jeff LaCoursiere]

Its not about email - the Voicemail app allows you to record a message 
that gets dropped into multiple voicemail boxes.


Thanks,

j

On 04/12/2017 12:15 PM, Telium Technical Support wrote:

Why not use an ALIAS and let sendmail send the email to a distribution
group?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, April 12, 2017 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] AGI Exec Voicemail


Hi,

I have a voicemail broadcast AGI that has been running fine for years - it
collects extensions and then EXECs the Voicemail app, like this:

EXEC Voicemail \"%s\"

(%s is the extension list like AAA etc)

This works fine, but after leaving the message and pressing "#", I just get
"Thank you" and a hangup.  I would like to have the option to review,
re-record, or cancel.  It isn't clear how to enable this option via EXEC.  I
tried:

EXEC Voicemail \"%s,review=yes\"

but there is no effect at all.

Any clues?

Thanks,

j

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[asterisk-users] AGI Exec Voicemail

[Jeff LaCoursiere]

Hi,

I have a voicemail broadcast AGI that has been running fine for years - 
it collects extensions and then EXECs the Voicemail app, like this:


EXEC Voicemail \"%s\"

(%s is the extension list like AAA etc)

This works fine, but after leaving the message and pressing "#", I just 
get "Thank you" and a hangup.  I would like to have the option to 
review, re-record, or cancel.  It isn't clear how to enable this option 
via EXEC.  I tried:


EXEC Voicemail \"%s,review=yes\"

but there is no effect at all.

Any clues?

Thanks,

j

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[asterisk-users] E-911

[Jeff LaCoursiere]

Apologies if this is considered off-topic; I suspect the information 
might benefit a portion of the list.


Can anyone point me in a direction to start implementation of E-911 
services?  Is this just something my upstream should supply, or can I 
connect to something on my own?


Thanks,

--
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312 962 5250 desk
815 546 6599 cell


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[asterisk-users] multiple outbound invites

[Jeff LaCoursiere]

Hello,

I have two upstream providers we use for US termination.  The dialplan 
sends calls out the "primary" and if that fails for specific reasons, it 
sends the same call out the "secondary". This has worked well for us 
when we are lazy about keeping balances up, for example.


Starting a few days ago ALL calls sent to the 'primary' were returned as 
busy, though the secondary terminated them fine.  We have a balance, and 
funny enough international calls are going through fine, just not US 
calls.  I opened a ticket.


The response form the carrier is that our asterisk is sending four 
simultaneous invites within one second, and for that reason the call is 
rejected.


I did a packet trace and was able to confirm this is true - only US 
calls sent to this carrier cause our end to send four identical 
simultaneous invites.  When it fails, a single invite for the same call 
is sent to the secondary, which is terminated without issue.


Happy to send the SIP trace if any would care to see it, but is there a 
reason anyone can think of that our asterisk (11.11.0) would suddenly 
start doing this?  It may be that it has been doing it all along, and 
our carrier just started rejected calls that come in this way, I'm not sure.


Cheers,

j


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[asterisk-users] inbound T38 to email

[Jeff LaCoursiere]

I have played around with iaxmodem and hylafax and have a few working 
installations where PRI's are involved.  I have a new customer that will 
be sending inbound fax calls via a new (SIP) DID provider we are working 
with (yup, same one from the last message I just sent), and they support 
T38.


I vaguely remembered a 't38modem' project on sourceforge and integration 
with hylafax, and started looking at that today, but t38modem hasn't 
been touched since 2009.


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


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312 962 5250 desk
815 546 6599 cell   


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[asterisk-users] new inbound DID provider... no auth?

[Jeff LaCoursiere]

We are trying to work with a new DID provider and I find myself 
confused.  Their standard integration is to send the call with no 
authentication.  I am expected to whitelist all their possible gateways, 
and accept their calls I guess with no peer definition.  I actually have 
it working this way; the calls land in our "public" context, I guess as 
"guest", and I am able to route them from there.  But that makes me nervous.


I would rather at least have them be associated with a defined peer, so 
I can set the right context and any other parameters I might want 
associated.  It is inbound only, no outbound.  I might try to set a 
host= in a peer definition with no secret, and see if that matches it, 
but I would rather avoid making a peer definition for every gateway they 
have.  Can anyone think of a way to define a single peer that might show 
from multiple potential addresses without authentication info?


Cheers,

--
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312 962 5250 desk
815 546 6599 cell   


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Re: [asterisk-users] Metaswitch caller ID passing

[Jeff LaCoursiere]

This did get resolved on the carrier side - thank to all who pointed me 
towards documentation and suggestions.  The Metaswitch has an option 
"Caller Number Connected Line ID Screening" which was set to "owned".  
Changing this to "valid format" allowed my asterisk instances to send 
foreign caller ID information and it was accepted.


Cheers,

j

On 08/05/2016 09:05 AM, Jeff LaCoursiere wrote:

Hi,

I am dealing with a telco that has recently upgraded from a DMS100 
switch to a "Metaswitch", and our PRI no longer passes foreign caller 
ID information, i.e. if I place an outbound call with specific caller 
ID information not associated with the PRI, it gets replaced with the 
PRI's primary phone number.


This is a bit of a problem for follow-me services, which end up 
showing the PRI's primary phone number instead of the original 
caller's phone number.


I know this isn't a "metaswitch" forum, but can anyone point me in a 
direction of some metaswitch documentation or know what the option is 
in a metaswitch to allow foreign caller ID information? The telco 
engineers are still struggling with this new switch, and I am not sure 
they understand or appreciate my urgency in getting this resolved!


Cheers,




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[asterisk-users] Metaswitch caller ID passing

[Jeff LaCoursiere]

Hi,

I am dealing with a telco that has recently upgraded from a DMS100 
switch to a "Metaswitch", and our PRI no longer passes foreign caller ID 
information, i.e. if I place an outbound call with specific caller ID 
information not associated with the PRI, it gets replaced with the PRI's 
primary phone number.


This is a bit of a problem for follow-me services, which end up showing 
the PRI's primary phone number instead of the original caller's phone 
number.


I know this isn't a "metaswitch" forum, but can anyone point me in a 
direction of some metaswitch documentation or know what the option is in 
a metaswitch to allow foreign caller ID information?  The telco 
engineers are still struggling with this new switch, and I am not sure 
they understand or appreciate my urgency in getting this resolved!


Cheers,

--
j


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Re: [asterisk-users] Asterisk and Yealink T21P E2

[Jeff LaCoursiere]

On 07/14/2016 02:14 PM, Marcelo Terres wrote:

Hello.

Anybody in the list is using this IP phone?

Regards,

Marcelo H. Terres <mhter...@gmail.com>
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres



Sure.  Tons of them.

--
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312 962 5250 desk
815 546 6599 cell


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Re: [asterisk-users] Asterisk Development Company in India

[Jeff LaCoursiere]

And punctuation and grammar skills have we too!  Our english be VERY good

On 03/31/2016 02:20 AM, ankur verma wrote:

Have you ever heard of Asterisk Development.There are only few companies in
India which are providing this service and "Anticlock Technologies is one of
them.it is dealing in this field from long time and We provide client
satisfaction, full support and long term services
.




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Re: [asterisk-users] load test docker images?

[Jeff LaCoursiere]

On 02/24/2016 04:49 PM, Steve Edwards wrote:

On Fri, 19 Feb 2016, Jeff LaCoursiere wrote:

Has anyone created any docker images I might be able to use on EC2 
for load testing an asterisk platform?  I started an instance this 
morning and was about to load sipp and other tools, and then thought 
surely someone must have done this already.  I'd like to hammer a 
platform we have created with multiple EC2 images until it breaks, to 
test capacity.


I'm surprised no one has this available.

If it's in your skill set, I could make use of it.



I did actually find a few docker images having to do with sipp, but all 
seemed to be basically a container with sipp loaded.  Not all that 
useful, as I had sipp built in about ten minutes.


I'm working on scenarios to do our load test, which at present will 
involve pjsua and sipp sending audio calls through our proxy.  I'm 
thinking each container will add on fifty simultaneous calls, and I will 
simply start containers until I can't make an audible call myself 
through the proxy.  Not the grandest plan I suppose.  Open for 
suggestions :)


When I am done I will happily make the container available to the list 
via our website.


Cheers,

j

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[asterisk-users] load test docker images?

[Jeff LaCoursiere]

Has anyone created any docker images I might be able to use on EC2 for 
load testing an asterisk platform?  I started an instance this morning 
and was about to load sipp and other tools, and then thought surely 
someone must have done this already.  I'd like to hammer a platform we 
have created with multiple EC2 images until it breaks, to test capacity.


Cheers,

j

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Re: [asterisk-users] 1000 analogue lines with asterisk

[Jeff LaCoursiere]

That would be the expensive route.  The inexpensive route would be to 
buy FXS ethernet gateways, like this: 
http://www.voipsupply.com/grandstream-gxw4248.  You could then get by 
with a single reasonably sized asterisk box (probably two setup as HA) 
and no need for expensive cards or complex channel bank setup.  We have 
done many hotels this way with great results.


j

On 02/17/2016 05:39 PM, chris wrote:


+1

spending money to get that many fxs ports is going to negate any 
savings of reusing analog phones instead of buying ip phones


1000 analog ports sounds like hell and if it was me I  would be 
embarrassed to have a setup like that tied to my name if I was a 
consultant etc. Someone will come in after you and ask who set it up 
and the customer will say you :)


On Feb 17, 2016 4:23 AM, "A J Stiles" > wrote:


On Wednesday 17 Feb 2016, Goke Aruna wrote:
> Hello all,
> Can someone recommend what hardware to use for a 1000 analogue line
> capacity asterisk PABX?
>
> Regards

A PCI express card with four primary rate ISDN ports, each linked
up to a
channel bank, will give you 120 analogue lines.  So you will need
nine such
cards; and for reasons of simple numbers of slots on a
motherboard, they will
have to be split among three or more servers, linked to a gigabit
switch.

You might end up getting a better deal if you bought 1000 hardware
SIP phones.
(You also would probably increase your personal indispensability
factor, into
the bargain .)

--
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] weather.agi

[Jeff LaCoursiere]

On 12/16/2015 11:24 AM, Ryan Crowder wrote:

http://www.wunderground.com/weather/api/


Awesome!  Perfect!

Cheers,

j



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
d...@donkelly.biz
Sent: Wednesday, December 16, 2015 9:20 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] weather.agi



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, December 16, 2015 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] weather.agi

Here is a funny story.  We mostly do hotels in the Caribbean, and one of our
first clients (going on ten years now) used the sample "weather.agi"
that used to be shipped with... asterisk@home? Trixbox?  Can't even recall
where we originally got it from.

This perl script uses festival to speak a brief weather forecast to the
caller.  We told our hotels this was a feature for guests, and assigned a
function code to execute it.  They have been using it for years.  In fact
the front desk used to just dial it in speaker mode for guests checking in.

Here is the funny part - the text for the weather forecast was coming from
an anonymous FTP site at tgftp.nws.noaa.gov.  Sometime in 2013 they stopped
updating the forecast!  So ever since then the caller would get the forecast
for June something 2013.  But because the weather in St Thomas is nearly the
same year round, NO ONE NOTICED!  Well, until just recently.

After we stopped laughing about it, they now want me to fix it.  But I can't
seem to find a new source of textual summary weather data.

So... has anyone else run into this and fixed it by chance?  Or can point me
to weather data?

Cheers,

j


Does something here work for you?
http://search.usa.gov/search?v%3Aproject=firstgov=web+service
e=nws.noaa.gov

   --Don

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[asterisk-users] weather.agi

[Jeff LaCoursiere]

Here is a funny story.  We mostly do hotels in the Caribbean, and one of 
our first clients (going on ten years now) used the sample "weather.agi" 
that used to be shipped with... asterisk@home? Trixbox?  Can't even 
recall where we originally got it from.


This perl script uses festival to speak a brief weather forecast to the 
caller.  We told our hotels this was a feature for guests, and assigned 
a function code to execute it.  They have been using it for years.  In 
fact the front desk used to just dial it in speaker mode for guests 
checking in.


Here is the funny part - the text for the weather forecast was coming 
from an anonymous FTP site at tgftp.nws.noaa.gov.  Sometime in 2013 they 
stopped updating the forecast!  So ever since then the caller would get 
the forecast for June something 2013.  But because the weather in St 
Thomas is nearly the same year round, NO ONE NOTICED!  Well, until just 
recently.


After we stopped laughing about it, they now want me to fix it.  But I 
can't seem to find a new source of textual summary weather data.


So... has anyone else run into this and fixed it by chance?  Or can 
point me to weather data?


Cheers,

j



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Re: [asterisk-users] Asterisk encrypted authentication for clients

[Jeff LaCoursiere]

On 10/29/2015 04:01 PM, Motty wrote:



On 10/29/2015 01:11 PM, Jeff LaCoursiere wrote:

On 10/28/2015 06:37 PM, Pete Mundy wrote:

Hi Motty,

Isn't the whole point of the nonce in a SIP registration to ensure 
the secret doesn't go on the wire in plain-text? Is this not enough, 
or are you looking to hide the username too?


(if so, fair 'nuf, just wondering why :)

Pete

Ps, if so then I think TLS is the missing part of your equation.

On 29/10/2015, at 11:54 AM, Motty <motty.c...@gmail.com> wrote:


Hello,
I am searching for a solution to encrypt authentication from 
Asterisk server to clients. Searching srtp seem to encrypt traffic, 
I just want client authentication with encryption. Can someone 
point to the right direction? has anybody used ZRTP? experience 
with ZRTP?


Thanks,
_motty





You want SIP over TLS.  That encrypts the signalling.  SRTP and ZRTP 
encrypt the actual voice traffic.


Cheers,

j





Thanks Jeff,
I don't want SIP over TLS. I would like to encrypt password only, I 
suppose over TLS.


Thanks,
_motty


The password isn't sent - SIP auth involves a challenge/response with 
hashing (digest authentication).  If that's all you are interested in, 
you are already there.


Cheers,

j

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Re: [asterisk-users] Asterisk encrypted authentication for clients

[Jeff LaCoursiere]

On 10/28/2015 06:37 PM, Pete Mundy wrote:

Hi Motty,

Isn't the whole point of the nonce in a SIP registration to ensure the 
secret doesn't go on the wire in plain-text? Is this not enough, or 
are you looking to hide the username too?


(if so, fair 'nuf, just wondering why :)

Pete

Ps, if so then I think TLS is the missing part of your equation.

On 29/10/2015, at 11:54 AM, Motty > wrote:



Hello,
I am searching for a solution to encrypt authentication from Asterisk 
server to clients. Searching srtp seem to encrypt traffic, I just 
want client authentication with encryption. Can someone point to the 
right direction? has anybody used ZRTP? experience with ZRTP?


Thanks,
_motty





You want SIP over TLS.  That encrypts the signalling.  SRTP and ZRTP 
encrypt the actual voice traffic.


Cheers,

j
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Re: [asterisk-users] Test

[Jeff LaCoursiere]

Fail.

On 10/28/2015 04:42 PM, ama...@sevana.fi wrote:


Hi,

Just checking if my emails reach the list.

Thanks,
Amanda





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[asterisk-users] SIP headers in outofcall messages

[Jeff LaCoursiere]

Hi,

Our custom application sets some SIP headers that we want passed to the 
called party via asterisk in a simple proxy setup.  It works fine for 
voice calls, but we also use SIP to send outofcall messages.  I notice I 
can't use SIP_HEADER() to get those custom SIP headers in outofcall 
messages.  Is this a bug?


I have this in sip.conf:

[general]
accept_outofcall_message=yes
outofcall_message_context=astsms

We can send and receive messages between extensions with this without 
issue.  In extensions.conf:


[astsms]
exten => _X.,1,Set(CHATNOSTORE=${SIP_HEADER(X-Semo-ChatNoStoreForward)})
exten => _X.,n,NoOp(CHATNOSTORE ${CHATNOSTORE})
exten => _X.,n,SIPAddHeader(X-Semo-ChatNoStoreForward: ${CHATNOSTORE})
[snip the parts that actually send on the message]


CHATNOSTORE gets set to "", even though I can see the header in a packet 
trace:



0x0260:  654f 7267 3a20 6865 6c6c 6f0d 0a58 2d53 eOrg:.hello..X-S
0x0270:  656d 6f2d 4368 6174 4e6f 5374 6f72 6546 emo-ChatNoStoreF
0x0280:  6f72 7761 7264 3a20 666f 7277 6172 645f orward:.forward_
0x0290:  7661 6c75 650d 0a0d 0a54 6573 7420 6d65 valueTest.me
0x02a0:  7373 6167 6520 31 ssage.1

I'm not sure where to take this next... dive into the code for SIP_HEADER?

Thanks,

j

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[asterisk-users] partially match a SIP header

[Jeff LaCoursiere]

Hi,

I have a need to pass through SIP headers that start with a particular 
prefix, without knowing beforehand what the full name of the header 
actually is.  For example I need to test for any headers on an inbound 
channel that start with "FOO_" and then use SIPAddHeader() to add them 
to the outbound channel.  Straightforward when I know the whole name, 
but I'd like something more generic that I won't have to update when we 
come up with a new header to add on the client side.


This is working fine:

exten => _X.,1,Set(FOO_ONE=${SIP_HEADER(FOO_ONE)})
exten => _X.,n,NoOp(Got Header ${FOO_ONE})
exten => _X.,n,SIPAddHeader(FOO_ONE: ${FOO_ONE})
exten => _X.,n,Dial(SIP/${EXTEN},80)
exten => _X.,n,Hangup


Any ideas?

Thanks,

j

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Re: [asterisk-users] Xorcom T1 to PRI

[Jeff LaCoursiere]

On 09/30/2015 07:47 AM, Tzafrir Cohen wrote:

[snip]
Right. For Sangoma cards, lsdahdi can't tell if the port is E1 or T1 and
thus calls it "PRI". Note that "PRI" here is a poor name that refers to
the port type itself and not to the signalling in it (which don't have
to be ISDN).

Suggestions? Patches?



Ah, well that explains it, and I don't need to worry that I have missed 
some config.  Might I suggest that lsdahdi show "E1/T1" if the actual 
type cannot be determined due to hardware limitations?


Cheers,

j

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[asterisk-users] Xorcom T1 to PRI

[Jeff LaCoursiere]

Hi,

I have a client that has a 24 channel voice T1 that I have been using 
e signalling over for a number of years.  The local telco finally got 
an ISDN switch and wants to move them to PRI.  I didn't see this as a 
big problem - I've done a few others on this particular Caribbean island 
without issues, but this would be the first time with a Xorcom unit 
involved.


We tried to do it tonight and failed.  I suspect there is an issue at 
their central office where they patched the line away from an old 
dms-100 and to the new ISDN switch - the tech said he couldn't see 
anything at the line layer, though I showed no alarms.  Regardless, 
something odd happened that I am hoping for some advice on.  We reverted 
to the T1 and will try again next week.


The odd bit is this - lsdahdi showed all 24 channels as "T1" instead of 
"PRI".  The output of lsdahdi at another site where I had already done 
this conversion (which uses a Sangoma card) looks like this:


root@astsouth:/etc/dahdi# lsdahdi
### Span  1: WPT1/0 "wanpipe1 card 0" (MASTER) ESF/B8ZS
  1 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  2 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  3 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  4 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  5 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  6 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  7 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  8 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
  9 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 10 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 11 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 12 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 13 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 14 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 15 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 16 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 17 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 18 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 19 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 20 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 21 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 22 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 23 PRIClear   (In use) (EC: WANPIPE_HWEC - INACTIVE)
 24 PRIHardware-assisted HDLC  (In use) (EC: WANPIPE_HWEC - 
INACTIVE)


Bah, the scrollback has already erased the output from the box in 
question, but essentially it showed the same as above with "T1" in the 
second column for all channels instead of "PRI".


They both had identical system.conf, which looks like this:

loadzone=us
defaultzone=us
span=1,1,0,esf,b8zs
bchan=1-23
echocanceller=mg2,1-23
hardhdlc=24

In fact the *only* things I changed from the old system.conf tonight was 
swapping "e=1-24" for "bchan=1-23" and adding the "hardhdlc=24".  The 
old (now functioning) system.conf looks like this:


# Span 1: XBUS-00/XPD-00 "Xorcom XPD [usb:X1067719].1: T1" (MASTER) 
B8ZS/ESF RED

loadzone= us
defaultzone= us
span=1,1,0,esf,b8zs
e=1-24
echocanceller=mg2,1-24

So what about system.conf would cause lsdahdi to show "T1" instead of 
"PRI" in column two?  Just trying to head off any additional problems 
once they get their patching sorted out.


Thanks,

j

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Re: [asterisk-users] Xorcom T1 to PRI

[Jeff LaCoursiere]

[snip]


So what about system.conf would cause lsdahdi to show "T1" instead of
"PRI" in column two?  Just trying to head off any additional problems
once they get their patching sorted out.



The issue is probably with the wanpipe configuration and not with 
DAHDI or Asterisk.  Run the wanpipe_cfg script again and make sure you 
change all necessary settings from E to ISDN PRI.




The site with the Sangoma (wanpipe) card is working as expected as a 
PRI.  The site that I couldn't switch tonight uses a Xorcom box - the 
only one I have worked on.  Is there also something about Xorcom config 
that needs to change to make it realize it is a PRI and not a T1?  I 
don't recall anything like that in the installation, but that was some 
time ago.


Thanks,

j


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[asterisk-users] Dell portability

[Jeff LaCoursiere]

Howdy,

I built an LXC container with an image of asterisk 11.18 precompiled 
and installed.  It runs fine on the dev platform, which is a Dell R320 
running Ubuntu 14.04LTS.  I shutdown the container, tarred it up, and 
untarred on a Dell PE1850, also running Ubuntu 14.04LTS.  The container 
itself is Ubuntu 14.04LTS.  Both platforms as far as I know are amd64.


The container boots fine on the 1850, but trying to run asterisk 
segfaults.  The source tree was still in the container, so I just did a 
make clean; make; make install.  It now runs fine.


Is there some compile flag I could use to make sure it is more 
compatible as I copy the container around?  Can anyone suggest a debug 
sequence that would at least narrow down what is causing the fault?


Cheers,

j

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Re: [asterisk-users] Dell portability

[Jeff LaCoursiere]

Awesome, that did the trick!  Thanks!

j

On 07/01/2015 01:44 PM, Scott Griepentrog wrote:
Try turning off BUILD_NATIVE in menuselect.  This will eliminate 
optimizations for the processor you last compiled on, which prevents 
crashes due to instructions not present on a different processor. This 
is frequently necessary when using in virtual environments.


In cli form:  # menuselect/menuselect --disable BUILD_NATIVE



On Wed, Jul 1, 2015 at 1:36 PM, Jeff LaCoursiere j...@jeff.net 
mailto:j...@jeff.net wrote:



Howdy,

I built an LXC container with an image of asterisk 11.18
precompiled and installed.  It runs fine on the dev platform,
which is a Dell R320 running Ubuntu 14.04LTS.  I shutdown the
container, tarred it up, and untarred on a Dell PE1850, also
running Ubuntu 14.04LTS.  The container itself is Ubuntu
14.04LTS.  Both platforms as far as I know are amd64.

The container boots fine on the 1850, but trying to run asterisk
segfaults.  The source tree was still in the container, so I just
did a make clean; make; make install. It now runs fine.

Is there some compile flag I could use to make sure it is more
compatible as I copy the container around?  Can anyone suggest a
debug sequence that would at least narrow down what is causing the
fault?

Cheers,

j

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Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org




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[asterisk-users] use of EC2

[Jeff LaCoursiere]

Curious if anyone has any stats on max concurrent calls on different EC2 
instance sizes.  A client has a proof of concept running on a medium 
compute instance now, and we are curious at what point we might 
experience issues.  All calls are SIP, no transcoding, using SPEEX.  I'd 
love to hear if anyone has a small or medium compute instance doing  
100 simultaneous calls.


Cheers,

j

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Re: [asterisk-users] OpenVZ with asterisk 13

[Jeff LaCoursiere]

On 04/07/2015 10:48 AM, Johan Wilfer wrote:

Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:

Dear all,

Is anyone has experience making Asterisk server with virtual server
OPEN-VZ (in proxmox 3.4 box) ?

My boss want to build a production server with it, and it will have +/-
300 sip user (concurrent call maybe  150 call)



As long as you don't overload the server it works great. I've used 
OpenVZ to separate Asterisk instances from each other. For my 
application (mostly conferencing) I can put ~ 350 concurrent calls on 
a single HP Xeon server.


OpenVZ is not really like KVM but more like Solaris containers or BSD 
jails. Docker is mostly using the same Kernel api:s that OpenVZ uses, 
but OpenVZ also has some cusom stuff.


If you need Dahdi you will need to give the VE's access to these 
devices, there are articles out there that explain how this is done.


Good luck!



We use LXC (what is under Docker) instead of OpenVZ to separate asterisk 
instances, and when Dahdi is needed I typically run an asterisk instance 
on the host and have SIP trunks between the container and the host 
instances.


Cheers,

j

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Re: [asterisk-users] RTP handling

[Jeff LaCoursiere]

On 03/24/2015 04:28 PM, Richard Mudgett wrote:



On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net 
mailto:j...@jeff.net wrote:



Hello,

I am wondering if asterisk does anything at all to RTP packets
passed from channel to channel if no transcoding is involved? Can
I assume that the packet that left phone A, arrived at the
asterisk server, was copied to phone B's channel and eventually
arrived at phone B had exactly (byte for byte) the same payload? 
Assume two SIP endpoints, no NAT involved.



That will only happen when the call is natively bridged:

Non-native bridge: Packets can get translated or Asterisk has an 
interest in the packet for things like DTMF or call recording.
Native bridge doing packet-to-packet (Local bridging): Packets come in 
on one channel and go out the other channel with nothing else done to 
them.
Native bridge doing direct media (Remote bridging): Packets go 
directly between endpoints so Asterisk never sees them.


Richard



Thanks for the quick reply RIchard!  Can I force native bridging, or 
does it default to that if I don't configure direct media?  The dialplan 
will be very simple - extensions calling extensions within a context.  
No DTMF, no recording, no mixing for conference, etc.


Cheers,

j

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[asterisk-users] RTP handling

[Jeff LaCoursiere]

Hello,

I am wondering if asterisk does anything at all to RTP packets passed 
from channel to channel if no transcoding is involved? Can I assume that 
the packet that left phone A, arrived at the asterisk server, was copied 
to phone B's channel and eventually arrived at phone B had exactly (byte 
for byte) the same payload?  Assume two SIP endpoints, no NAT involved.


Thanks,

j

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Re: [asterisk-users] PRI Callerid Passthrough

[Jeff LaCoursiere]

My, how embarrassing.  I of course meant that as a personal message to 
Don.  But if anyone else knows the answer, I'm interested! lol


Cheers,

j

On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote:


Hey Don,

How are you?  I may be heading your way in the next month or so.  Have 
to meet with a guy in Eden Prairie, and stop off at my 
brother/sisterm-in-law's as well.


Got a question for you - with TBCT, who pays for the call once it is 
transferred?  Still me as the owner of the trunk?


Lets say I take a call that was dialled locally (caller believes this 
is free), and I do a TBCT to an international destination, and they 
stay on the line for ten minutes.  Who gets the bill?


Cheers,

j

On 03/18/2015 09:19 AM, d...@donkelly.biz wrote:


This depends on what you mean by “not involving the service provider.”

If you are literally forwarding calls that come in on the PRI back 
out on the PRI, the most efficient way is with Two B-Channel Transfer 
(TBCT). Check it out in the wiki.


You need to make sure your carrier supports the feature.

When you want to do a “transfer,” you have an incoming call alerting 
or answered, you initiate an outgoing call (using the originating 
ANI). You initiate the TBCT and the CARRIER completes the transfer, 
disconnecting both of your B channels. The carrier will later notify 
you when the transferred call is done, but I don’t think Asterisk 
handles this directly.


Note that at least one of the calls must be answered when you 
initiate the transfer. If you are doing “unattended” transfer, you 
will, typically, leave the incoming call alerting until the outbound 
call answers, then complete the transfer. An “attended” transfer 
would generally answer the incoming call, play a message, do some IVR 
doodling, or chat with an agent then initiate the transfer.


Have fun

--Don

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Rizwan H Qureshi

*Sent:* Wednesday, March 18, 2015 7:16 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PRI Callerid Passthrough

Thanks AJ and David,

We were actually using GSM gateways by setting busy forward number on 
the SIMs and just giving busy signal on every incoming call, telco 
took care of the forwarding and the line was free within seconds. Now 
we need to scale up the setup but GSM gateways a very very expensive 
if we want to scale upto a 1000 DIDs, which means thousand SIMs and a 
gateway/gateways big enough.


On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles 
asterisk_l...@earthshod.co.uk 
mailto:asterisk_l...@earthshod.co.uk wrote:


On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote:
 Hi All,
 I have to forward incoming call on PRI back out to PRI but I need the
 original Callerid to passthrough. Is it possible with DAHDI PRI cards
 without involving the service provider?

 Thanks

It depends who your service provider is!

Any PRI card can send the commands down the D-channel to set any 
caller ID you

like, but it's still up to the telco whether or not they will honour your
request.  I know the hard way that BT will only let you identify with 
a number

you're entitled to use.

Also, remember if you have a call coming in on a PRI line and going 
out on

another PRI line, that's eating two of your thirty lines .

--
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] PRI Callerid Passthrough

[Jeff LaCoursiere]

Hey Don,

How are you?  I may be heading your way in the next month or so. Have to 
meet with a guy in Eden Prairie, and stop off at my 
brother/sisterm-in-law's as well.


Got a question for you - with TBCT, who pays for the call once it is 
transferred?  Still me as the owner of the trunk?


Lets say I take a call that was dialled locally (caller believes this is 
free), and I do a TBCT to an international destination, and they stay 
on the line for ten minutes.  Who gets the bill?


Cheers,

j

On 03/18/2015 09:19 AM, d...@donkelly.biz wrote:


This depends on what you mean by “not involving the service provider.”

If you are literally forwarding calls that come in on the PRI back out 
on the PRI, the most efficient way is with Two B-Channel Transfer 
(TBCT). Check it out in the wiki.


You need to make sure your carrier supports the feature.

When you want to do a “transfer,” you have an incoming call alerting 
or answered, you initiate an outgoing call (using the originating 
ANI). You initiate the TBCT and the CARRIER completes the transfer, 
disconnecting both of your B channels. The carrier will later notify 
you when the transferred call is done, but I don’t think Asterisk 
handles this directly.


Note that at least one of the calls must be answered when you initiate 
the transfer. If you are doing “unattended” transfer, you will, 
typically, leave the incoming call alerting until the outbound call 
answers, then complete the transfer. An “attended” transfer would 
generally answer the incoming call, play a message, do some IVR 
doodling, or chat with an agent then initiate the transfer.


Have fun

--Don

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rizwan 
H Qureshi

*Sent:* Wednesday, March 18, 2015 7:16 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PRI Callerid Passthrough

Thanks AJ and David,

We were actually using GSM gateways by setting busy forward number on 
the SIMs and just giving busy signal on every incoming call, telco 
took care of the forwarding and the line was free within seconds. Now 
we need to scale up the setup but GSM gateways a very very expensive 
if we want to scale upto a 1000 DIDs, which means thousand SIMs and a 
gateway/gateways big enough.


On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles 
asterisk_l...@earthshod.co.uk mailto:asterisk_l...@earthshod.co.uk 
wrote:


On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote:
 Hi All,
 I have to forward incoming call on PRI back out to PRI but I need the
 original Callerid to passthrough. Is it possible with DAHDI PRI cards
 without involving the service provider?

 Thanks

It depends who your service provider is!

Any PRI card can send the commands down the D-channel to set any 
caller ID you

like, but it's still up to the telco whether or not they will honour your
request.  I know the hard way that BT will only let you identify with 
a number

you're entitled to use.

Also, remember if you have a call coming in on a PRI line and going out on
another PRI line, that's eating two of your thirty lines .

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying 
off-

list, change address to asterisk1list at earthshod dot co dot uk .


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[asterisk-users] DND on a Polycom IP450

[Jeff LaCoursiere]

Only slightly asterisk related I suppose, but hoping someone has 
attempted this...


I have an old installation with a bunch of IP501s, and one died.  I 
replaced it with an IP450, and the user sorely misses his DND button.  I 
hated those DND buttons anyway, as I couldn't control them centrally.


I'd *like* to program one of his softkeys to send a *XX sequence to do 
DND on the asterisk-side instead of on the phone.  Has anyone remapped 
the softkeys on an IP450 this way and might share the XML?


Thanks!

j

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Re: [asterisk-users] AWS/EC2 server selection

[Jeff LaCoursiere]

Amazon instances are shared resources.  I wouldn't want to count on 
timing or disk throughput, and you can't just ask them to do ssd - its 
a virtual machine!  500 simultaneous recordings is a hefty load, and I 
would want to know that the underlying hardware is dedicated to the task.


Sure you see lots of posts about hosting asterisk and/or freeswitch on 
EC2.  I have done it myself and even have some clients doing it now *for 
proof of concept*.   I've never heard of anyone using it for the kind of 
load you are talking about.  I'm assuming with such a giant load you are 
making a decent profit. Buy some hefty hardware and do the architecture 
properly.  You can rent half a rack at lots of high end datacenters for 
less than $1000/month.


j

On 03/07/2015 12:43 AM, Amit Patkar wrote:

Hi Jeff

Are you aware of any challenges of hosting it on AWS? It will help me 
to work out alternate plan. Is there any recommendation? Should I 
split it to multiple instances and balance traffic across multiple 
small server instances? I can use Kamailio to balance traffic.


I see many posts referring to AWS deployment. Please help me to choose 
AWS server instance.


*Thanks  Regards,*
Amit Patkar


On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote:


Why use Amazon?  With that kind of load I would want dedicated 
servers.  Call Rackspace or Softlayer.


j

On 03/06/2015 11:59 AM, Amit Patkar wrote:

Hi

I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 + 
g.711) and full recording. Number of concurrent calls expected are 
500+. 2 instances will be configured for 100% redundancy. Heart beat 
will be used to determine active instance.

How should I choose EC2 instance?
How many vCPU, RAM should be selected? I am assuming that server 
with ssd is required as all 500+ calls needs to be recorded.


Regards,
Amit Patkar







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Re: [asterisk-users] AWS/EC2 server selection

[Jeff LaCoursiere]

Still a shared resource.  I don't see the benefit.

Even beyond the shared resource bit, with the kind of IO you are likely 
to be pushing, you will want a decent NAS with lots of spindles and 
fibre channel to your hosts.


j

On 03/08/2015 10:51 AM, Jai Rangi wrote:

Digital ocean offers ssd on all the virtual machines. Uptime is good.

Jai Rangi
Www.didforsale.com http://Www.didforsale.com
www.cebodtelecom.com http://www.cebodtelecom.com
www.cebod.com http://www.cebod.com

On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere j...@jeff.net 
mailto:j...@jeff.net wrote:




Amazon instances are shared resources.  I wouldn't want to count on 
timing or disk throughput, and you can't just ask them to do ssd - 
its a virtual machine!  500 simultaneous recordings is a hefty load, 
and I would want to know that the underlying hardware is dedicated to 
the task.


Sure you see lots of posts about hosting asterisk and/or freeswitch 
on EC2.  I have done it myself and even have some clients doing it 
now *for proof of concept*.   I've never heard of anyone using it for 
the kind of load you are talking about.  I'm assuming with such a 
giant load you are making a decent profit.  Buy some hefty hardware 
and do the architecture properly.  You can rent half a rack at lots 
of high end datacenters for less than $1000/month.


j

On 03/07/2015 12:43 AM, Amit Patkar wrote:

Hi Jeff

Are you aware of any challenges of hosting it on AWS? It will help 
me to work out alternate plan. Is there any recommendation? Should I 
split it to multiple instances and balance traffic across multiple 
small server instances? I can use Kamailio to balance traffic.


I see many posts referring to AWS deployment. Please help me to 
choose AWS server instance.


*Thanks  Regards,*
Amit Patkar


On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote:


Why use Amazon?  With that kind of load I would want dedicated 
servers.  Call Rackspace or Softlayer.


j

On 03/06/2015 11:59 AM, Amit Patkar wrote:

Hi

I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 + 
g.711) and full recording. Number of concurrent calls expected are 
500+. 2 instances will be configured for 100% redundancy. Heart 
beat will be used to determine active instance.

How should I choose EC2 instance?
How many vCPU, RAM should be selected? I am assuming that server 
with ssd is required as all 500+ calls needs to be recorded.


Regards,
Amit Patkar







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Re: [asterisk-users] AWS/EC2 server selection

[Jeff LaCoursiere]

Why use Amazon?  With that kind of load I would want dedicated servers.  
Call Rackspace or Softlayer.


j

On 03/06/2015 11:59 AM, Amit Patkar wrote:

Hi

I plan to host Asterisk instances on AWS/EC2 servers.
Requirement is to run asterisk instance with transcoding (g.729 + 
g.711) and full recording. Number of concurrent calls expected are 
500+. 2 instances will be configured for 100% redundancy. Heart beat 
will be used to determine active instance.

How should I choose EC2 instance?
How many vCPU, RAM should be selected? I am assuming that server with 
ssd is required as all 500+ calls needs to be recorded.


Regards,
Amit Patkar




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Re: [asterisk-users] [OT] switches

[Jeff LaCoursiere]

On 02/25/2015 09:28 AM, Steve Edwards wrote:

On Wed, 25 Feb 2015, A J Stiles wrote:

The limiting factor with a switch carrying IP telephony traffic is 
not bandwidth, but routing table entries; and even cheap switches 
nowadays will usually take 1024 entries, if not 4096.


Are you referring to the MAC CAM table? Saying 'routing table' and 
'switch' in the same sentence seems confusing.


Do VOIP devices take more table entries than other Ethernet devices? 
I.e. more than 1?




No, and if you have 1024 MAC addresses behind a cheap switch, you get 
what you deserve.


j

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Re: [asterisk-users] Astricom 2014 presentations

[Jeff LaCoursiere]

On 10/29/2014 05:50 AM, Bogdan Cristea wrote:

Hi

Will the presentations made at Astricom 2014 be made public as recorded videos ?

thanks
Bogdan


I'll second the request for that, and also ask if the sessions on 
Kamailio will be similarly available.


Cheers,

j

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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Jeff LaCoursiere]

On 09/23/2014 10:53 PM, Don Kelly wrote:

On Tue, 23 Sep 2014, Steve Edwards wrote:


  On 09/23/2014 02:17 PM, Steve Edwards wrote:

  For some applications, storing recorded audio (prompts and caller
  recordings) as a BLOB in MySQL has advantages.

On Tue, 23 Sep 2014, Don Kelly wrote:


  I'm curious about what the advantages are of storing audio in a blob.
  Wouldn't it be more efficient to store it in a file and just put the
filename in the database?

Multiple web servers, multiple Asterisk servers, multiple DB servers,
synchronizing filesystems vs db, etc.

It appears to eliminate some problems, but Asterisk limiting audio
playback to files seems like a tough obstacle.



Mike said:
Maybe make the audio files available to all servers via a single NFS
directory?  Probably not a good solution if the servers aren't co-located.


Maybe someone could write a Linux device file that would return the blob's
content as a file read.




I beat you to that one ;)  That is exactly what a named pipe (fifo) is.  
Asterisk would read it like a sound file, and the AGI would dump the 
BLOB to it on demand.  It would work, but you can't have more than one 
process at a time reading from it, so that's a further complication...


j

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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Jeff LaCoursiere]

On 09/23/2014 02:17 PM, Steve Edwards wrote:
For some applications, storing recorded audio (prompts and caller 
recordings) as a BLOB in MySQL has advantages.


So, once I have the audio in the database, how can I play it?

Creating temporary files seems so tacky.

Is there another way to playback or background audio either by 
specifying a URL or from a memory buffer (either C or PHP)?




How about a named pipe (fifo)?  Of course then you might have issues 
with simultaneous calls.  You would have to have a pool of them and 
somehow manage locking them...


j

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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Jeff LaCoursiere]

On 09/02/2014 03:14 PM, Administrator TOOTAI wrote:

Le 02/09/2014 20:18, Khalid Touati a écrit :

so it seems Asterisk Versions does not support video I guess


Asterisk supports video. I'm using it with asterisk 1.4 1.8 and 11 
with GrandStream phones (H263, H263+ and H264). Works perfectly




I can second that with GS phones, asterisk 1.4 and 1.8.

j


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Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

[Jeff LaCoursiere]

Don't forget videosupport=yes in sip.conf.

j


On 09/02/2014 03:52 PM, Eric Wieling wrote:

A co-worker was doing video, I dislike video.  The phones were Polycom VVX, The 
settings on our FreePBX box (office PBX) on the Settings / Asterisk SIP 
Settings / Video section we have Video: Enabled, H.263 and H.263p are the only 
two video codecs enabled.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 7:54 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 21:15, Eric Wieling wrote:

As long as you are NOT transcoding video should work in Asterisk.

Both apps were configured with identical (codec) settings so I don't see
how it would require transcoding. If you did get it to work I would
appreciate it if you could tell me which clients you used, the Asterisk
version, the OS and the relevant Asterisk config.

Thanks,
Patrick



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock
Sent: Tuesday, September 02, 2014 6:39 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

On 02-09-14 20:18, Khalid Touati wrote:

so it seems Asterisk Versions does not support video I guess

On a local LAN/Wifi I tried it briefly with Asterisk 11.12.0 and the
Bria app on Android and iPhone. With SELinux and the firewall
temporarily disabled I couldn't get it to work with either H264 or VP8.

HTH,
Patrick






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Re: [asterisk-users] OT: Question on Caller ID (Spoofing calls with Asterisk)

[Jeff LaCoursiere]

On 08/26/2014 09:55 AM, Doug Lytle wrote:

What I found curious was the caller's name was Asterisk

On our systems, if I don't assign a CID number to an inbound call that is 
blocking it's CID, the default shown on the Polycom phones is Asterisk.  I've 
set it up that any inbound call with no CID is assigned a 0 for the phone 
number and Restricted as the CID name.

Doug



So, in other words, they had a moron install the system that is trying 
hard to social engineer their way into people's computers.


j

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Re: [asterisk-users] PRI timing settings

[Jeff LaCoursiere]

On 08/20/2014 07:58 AM, Scott L. Lykens wrote:


On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net 
mailto:j...@jeff.net wrote:


I wrote earlier today about a new PRI installation in the Caribbean, 
where all outbound calls are functioning fine *except* calls to 
Sprint phone numbers, which get rejected immediately as busy.


I don’t know what expectations for CLID your carrier might have, or 
for that matter the upstream carrier, however, we found through our 
CLEC here in the US that while the CLEC was happy to take e.164 
formatted numbers from us as CLID, Global Crossing would reject them 
further upstream resulting in our calls to many toll frees being rejected.


Switching to 10 digit CLID on all outbound calls through that PRI 
solved the problem.


I don’t know if this is your problem but be sure your CLID is in the 
most simple format possible for your region to help rule it out.


sl



This makes me curious... what *is* the simplest format possible for 
NANPA numbers?  I'm sure there must be a spec to conform to.  Can anyone 
point me to it?


Cheers,

j
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Re: [asterisk-users] PRI timing settings

[Jeff LaCoursiere]

What about the text portion?  Should that never be sent?  I was indeed 
sending the '1', and I will remove that to see if it solves my problem, 
but I also have the company name in there.  I feel like a newb asking 
such questions, but I've never had this issue before :)


Company 1NXXNXX

Cheers,

j

On 08/20/2014 09:46 AM, Eric Wieling wrote:


NXXNXX is the correct format of CallerID numbers in NANPA.   The 
leading 1 is not part of any NANPA phone number.   Toll free “area 
codes” are also not valid for CallerID.


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff 
LaCoursiere

*Sent:* Wednesday, August 20, 2014 2:41 PM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings

On 08/20/2014 07:58 AM, Scott L. Lykens wrote:

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net
mailto:j...@jeff.net wrote:



I wrote earlier today about a new PRI installation in the
Caribbean, where all outbound calls are functioning fine *except*
calls to Sprint phone numbers, which get rejected immediately as
busy.

I don’t know what expectations for CLID your carrier might have,
or for that matter the upstream carrier, however, we found through
our CLEC here in the US that while the CLEC was happy to take
e.164 formatted numbers from us as CLID, Global Crossing would
reject them further upstream resulting in our calls to many toll
frees being rejected.

Switching to 10 digit CLID on all outbound calls through that PRI
solved the problem.

I don’t know if this is your problem but be sure your CLID is in
the most simple format possible for your region to help rule it out.

sl


This makes me curious... what *is* the simplest format possible for 
NANPA numbers?  I'm sure there must be a spec to conform to.  Can 
anyone point me to it?


Cheers,

j





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Re: [asterisk-users] PRI timing settings

[Jeff LaCoursiere]

Sadly none of these changes have made any difference.  I'll report the 
resolution for posterity once we find it.


Thanks,

j

On 08/20/2014 10:13 AM, Don Kelly wrote:


It’s possible that Sprint is burping on the name. Try first dropping 
the “1.”  Then try dropping the name also, if necessary.


--Don

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jeff 
LaCoursiere

*Sent:* Wednesday, August 20, 2014 10:03 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings


What about the text portion?  Should that never be sent?  I was indeed 
sending the '1', and I will remove that to see if it solves my 
problem, but I also have the company name in there.  I feel like a 
newb asking such questions, but I've never had this issue before :)


Company 1NXXNXX

Cheers,

j

On 08/20/2014 09:46 AM, Eric Wieling wrote:

NXXNXX is the correct format of CallerID numbers in NANPA. The
leading 1 is not part of any NANPA phone number. Toll free “area
codes” are also not valid for CallerID.

*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jeff LaCoursiere
*Sent:* Wednesday, August 20, 2014 2:41 PM
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] PRI timing settings

On 08/20/2014 07:58 AM, Scott L. Lykens wrote:

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net
mailto:j...@jeff.net wrote:




I wrote earlier today about a new PRI installation in the
Caribbean, where all outbound calls are functioning fine
*except* calls to Sprint phone numbers, which get rejected
immediately as busy.

I don’t know what expectations for CLID your carrier might
have, or for that matter the upstream carrier, however, we
found through our CLEC here in the US that while the CLEC was
happy to take e.164 formatted numbers from us as CLID, Global
Crossing would reject them further upstream resulting in our
calls to many toll frees being rejected.

Switching to 10 digit CLID on all outbound calls through that
PRI solved the problem.

I don’t know if this is your problem but be sure your CLID is
in the most simple format possible for your region to help
rule it out.

sl


This makes me curious... what *is* the simplest format possible
for NANPA numbers?  I'm sure there must be a spec to conform to. 
Can anyone point me to it?


Cheers,

j







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Re: [asterisk-users] PRI timing settings

[Jeff LaCoursiere]

On 08/20/2014 12:04 PM, Andres wrote:

On 8/20/14, 11:28 AM, Steve Totaro wrote:

PRI intense debug should show all you need to fix this.

Right, the sooner you post this debug here the sooner we can help.  
Otherwise its just guesswork.


On Wed, Aug 20, 2014 at 12:13 PM, Jeff LaCoursiere j...@jeff.net 
mailto:j...@jeff.net wrote:



Sadly none of these changes have made any difference. I'll report
the resolution for posterity once we find it.

Thanks,

j





Ok, here is an intense debug trace.  I've replaced the phone numbers to 
protect the innocent.  The smoking gun seems to be this:


Ext: 1  Cause: Destination out of order (27)

Though I have no idea why... calling the same destination from my cell 
phone works fine.  We only send seven digits for local on-island calls 
like this, and calls to other carriers work fine with the same format.  
I'm starting to doubt there is anything I can do to fix this... seems 
like an issue between my telco and Sprint?


Cheers,

j

astsouth*CLI pri intense debug span 1
Enabled debugging on span 1
PRI Span: 1 t203_expire
PRI Span: 1
PRI Span: 1  TEI: 0 State 7(Multi-frame established)
PRI Span: 1  V(A)=17, V(S)=17, V(R)=73
PRI Span: 1  K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1  T200_id=0, N200=3, T203_id=0
PRI Span: 1  [ 00 01 01 93 ]
PRI Span: 1  Supervisory frame:
PRI Span: 1  SAPI: 00  C/R: 0 EA: 0
PRI Span: 1   TEI: 000EA: 1
PRI Span: 1  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1  N(R): 073 P/F: 1
PRI Span: 1  0 bytes of data
PRI Span: 1 -- Starting T200 timer
PRI Span: 1
PRI Span: 1  TEI: 0 State 8(Timer recovery)
PRI Span: 1  V(A)=17, V(S)=17, V(R)=73
PRI Span: 1  K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1  T200_id=8192, N200=3, T203_id=0
PRI Span: 1  [ 02 01 01 23 ]
PRI Span: 1  Supervisory frame:
PRI Span: 1  SAPI: 00  C/R: 1 EA: 0
PRI Span: 1   TEI: 000EA: 1
PRI Span: 1  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1  N(R): 017 P/F: 1
PRI Span: 1  0 bytes of data
PRI Span: 1
PRI Span: 1  TEI: 0 State 8(Timer recovery)
PRI Span: 1  V(A)=17, V(S)=17, V(R)=73
PRI Span: 1  K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1  T200_id=8192, N200=3, T203_id=0
PRI Span: 1  [ 02 01 01 93 ]
PRI Span: 1  Supervisory frame:
PRI Span: 1  SAPI: 00  C/R: 1 EA: 0
PRI Span: 1   TEI: 000EA: 1
PRI Span: 1  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1  N(R): 073 P/F: 1
PRI Span: 1  0 bytes of data
PRI Span: 1 -- Got ACK for N(S)=17 to (but not including) N(S)=17
PRI Span: 1 Done handling message for SAPI/TEI=0/0
PRI Span: 1
PRI Span: 1  TEI: 0 State 8(Timer recovery)
PRI Span: 1  V(A)=17, V(S)=17, V(R)=73
PRI Span: 1  K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1  T200_id=8192, N200=3, T203_id=0
PRI Span: 1  [ 00 01 01 23 ]
PRI Span: 1  Supervisory frame:
PRI Span: 1  SAPI: 00  C/R: 0 EA: 0
PRI Span: 1   TEI: 000EA: 1
PRI Span: 1  Zero: 0 S: 0 01: 1  [ RR (receive ready) ]
PRI Span: 1  N(R): 017 P/F: 1
PRI Span: 1  0 bytes of data
PRI Span: 1 -- Got ACK for N(S)=17 to (but not including) N(S)=17
PRI Span: 1 -- Stopping T200 timer
PRI Span: 1 -- Starting T203 timer
PRI Span: 1 Done handling message for SAPI/TEI=0/0
  == Using SIP RTP CoS mark 5
-- Executing [998@business:1] Dial(SIP/bolongo-1c78, 
DAHDI/g0/998,60) in new stack

PRI Span: 1 -- Making new call for cref 32897
-- Requested transfer capability: 0x00 - SPEECH
PRI Span: 1
PRI Span: 1  DL-DATA request
PRI Span: 1  Protocol Discriminator: Q.931 (8)  len=56
PRI Span: 1  TEI=0 Call Ref: len= 2 (reference 129/0x81) (Sent from 
originator)

PRI Span: 1  Message Type: SETUP (5)
PRI Span: 1 TEI=0 Transmitting N(S)=17, window is open V(A)=17 K=7
PRI Span: 1
PRI Span: 1  TEI: 0 State 7(Multi-frame established)
PRI Span: 1  V(A)=17, V(S)=17, V(R)=73
PRI Span: 1  K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
PRI Span: 1  T200_id=0, N200=3, T203_id=8192
PRI Span: 1  [ 00 01 22 92 08 02 00 81 05 04 03 80 90 a2 18 03 a1 83 81 
1e 02 80 83 28 0b b1 33 34 30 37 37 35 31 38 30 30 6c 0c 21 80 33 34 30 
37 37 35 31 38 30 30 70 08 80 39 39 38 39 39 36 35 ]

PRI Span: 1  Informational frame:
PRI Span: 1  SAPI: 00  C/R: 0 EA: 0
PRI Span: 1   TEI: 000EA: 1
PRI Span: 1  N(S): 017   0: 0
PRI Span: 1  N(R): 073   P: 0
PRI Span: 1  56 bytes of data
PRI Span: 1  Protocol Discriminator: Q.931 (8)  len=56
PRI Span: 1  TEI=0 Call Ref: len= 2 (reference 129/0x81) (Sent from 
originator)

PRI Span: 1  Message Type: SETUP (5)
PRI Span: 1  [04 03 80 90 a2]
PRI Span: 1  Bearer Capability (len= 5) [ Ext: 1  Coding-Std: 0 Info 
transfer capability: Speech (0)
PRI Span: 1   Ext: 1  Trans mode/rate: 
64kbps, circuit-mode (16)
PRI Span: 1 User information layer 1: 
u-Law (34)

PRI Span: 1  [18 03 a1 83 81]
PRI Span: 1  Channel ID (len= 5) [ Ext: 1  IntID: Implicit Other(PRI)  
Spare: 0