Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

[Bruce Ferrell]

On 6/14/20 4:52 PM, Antony Stone wrote:

On Monday 15 June 2020 at 00:41:14, Bruce Ferrell wrote:


Way back in the mists of time, I built my asterisk installation with SNMP
support.

Heh... I never even knew that was possible :)


That said, I actually prefer ARA/ARI to flat file configuration of endpoints
and dialplans.  Changes are more or less instantaneous and easily shared
between instances.

Agreed - ARA is a great system, and I really like that it can be combined with
flatfile configs on a single server.


The ODBC way is a pain, so I tend to just use the native MySQL method

Oh?

What makes you say ODBC is a pain?  I have two files (/etc/odbc.ini and
/etc/idbcinst.ini, which are 8 lines and 3 lines in size respectively) and I
had to install one file /usr/local/lib/libmaodbc.so to make it work with
MariaDB.


ODBC can get fussy about language definitions between the DB and ODBC.  ODBC make's it so you can use ANY db, but try Informix or Oracle... Just LOADS of fun getting the language 
code page settings right (Yes, I actually had a customer want Informix - SHUDDER).


The MySQL connection has it's own configuration fuss, but I understand it and 
HATE trying to debug ODBC configs.



What's easier about the native MySQL method?


for ARA configuration as well as CDR collection.  CDR reports are just a SQL
query away.

Yes:

a) efficient

b) can be done on a machien remote from the call processing

c) is realtime - call ends, CDRs are immediately available for analysis

d) can even include triggers on DB updates, for example to raise anti-fraud
alerts.


YES!  and most non-MySQL/Marias type don't realize that those triggers and 
stored procedures can actually execute routines external to the DB... At least 
in MySQL/MariaDB



Regards,


Antony.





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Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show calls"?

[Bruce Ferrell]

Way back in the mists of time, I built my asterisk installation with SNMP 
support.

It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that 
moment in time.


That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less instantaneous and easily shared between instances. The ODBC 
way is a pain, so I tend to just use the native MySQL method for ARA configuration as well as CDR collection.  CDR reports are just a SQL query away.


Only problem is all of the web ui's for Asterisk seem to be PHP that presumes 
the flatfiles.



On 6/14/20 2:55 PM, Telium Technical Support wrote:


Just run ‘core show calls’ as a command  from the AMI, and parse the results.  
I don’t think there is an equivalent pure AMI command.

*From:*asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On 
Behalf Of *Jonathan H
*Sent:* Sunday, June 14, 2020 5:45 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion 

*Subject:* Re: [asterisk-users] Any api (agi/ari/ami) equivalent of "core show 
calls"?

Wow! I've been *-ing for about 6 years and had literally no idea about that!

I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit 
hacky) dialplan.


That's making me sound ungrateful. I don't mean to be!

On Sun, 14 Jun 2020, 22:39 Steve Edwards, mailto:asterisk@sedwards.com>> wrote:

On Sun, 14 Jun 2020, Jonathan H wrote:

> Thank you... but "just update the database" - hmm, what database?

I used MySQL.

> Did you mean ARI? I still can't find the command! The asterisk wiki is
> somewhat, um... spread around!

ARA as in Asterisk RealTime Architecture

https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
https://www.voip-info.org/asterisk-realtime/

As I recall (back from 2015), you tell Asterisk which 'configuration file'
you want to read from MySQL like this:

# /etc/asterisk/extconfig.conf

[settings]
         musiconhold.conf                = mysql,vchat,static
;       musiconhold.conf                = mysql,vchat,musiconhold

I have no idea if this will help, but here are the tables as I defined them 
back in 2015.

         create  table   if not exists   static
                 (
                   id                    int(11) not null auto_increment
                 , cat_metric            int(11) not null default '0'
                 , var_metric            int(11) not null default '0'
                 , commented             int(11) not null default '0'
                 , filename              varchar(128) not null default ''
                 , category              varchar(128) not null default 
'default'
                 , var_name              varchar(128) not null default ''
                 , var_val               varchar(128) not null default ''
                 , primary key           (id)
                 )
                 ;

-- defaults
         set @CAT_METRIC                 = 0;
         set @FILENAME                   = 'musiconhold.conf';
         set @VAR_METRIC                 = 0;

-- Funk Dance
         set @COMMENTED                  = 0;
         set @NAME                       = 'Funk Dance';
         insert into static set
                   cat_metric            = @CAT_METRIC
                 , category              = @NAME
                 , commented             = @COMMENTED
                 , filename              = @FILENAME
                 , var_metric            = @VAR_METRIC
                 , var_name              = 'directory'
                 , var_val               = 
concat('/source/src/tmp/T2/moh/', @NAME, '/')
                 ;
         insert into static set
                   cat_metric            = @CAT_METRIC
                 , category              = @NAME
                 , commented             = @COMMENTED
                 , filename              = @FILENAME
                 , var_metric            = @VAR_METRIC
                 , var_name              = 'mode'
                 , var_val               = 'files'
                 ;
         insert into static set
                   cat_metric            = @CAT_METRIC
                 , category              = @NAME
                 , commented             = @COMMENTED
                 , filename              = @FILENAME
                 , var_metric            = @VAR_METRIC
                 , var_name              = 'sort'
                 , var_val               = 'random'
                 ;
         insert into static set
           

Re: [asterisk-users] Best way to update ever changing dialplans

[Bruce Ferrell]

On 06/25/2018 09:54 AM, Dovid Bender wrote:
I am working on a system where I connect to an external API and based on what it gives me I generate the Asterisk dial plan accordingly. I am thinking about my different options 
and wanted feedback from others on how to best do it.
1) Generate conf files for Asterisk - This seems the easiest but then I will be doing a dial plan reload on all of my dial plan for handful of lines of code. The plus side is once 
reload is don the dial plan is in memory.

2) Using real time + mysql - Seems like an overkill to have mysql running 
taking resources for a few lines.
3) Using real time + sqlite3 - This seems like the best option but then we go 
to disk every time there is a call.

Any other options that I am not thinking of?



I don't think you missed any options.

Call me silly, number 2 has always been my preference.  Flat file configuration 
bugs me except for the simplest
of things.

Since you mention the external API, you already have a fairly complex 
environment... And pretty likely to have, at
some point, multiple asterisk instances.  Surprise! The realtime mysql dialplan 
can be shared among them.

But I'm crazy that way.


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Re: [asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 4/25/18 6:05 AM, Joshua Colp wrote:

On Wed, Apr 25, 2018, at 12:40 AM, Bruce Ferrell wrote:




OK, I've gone back and refreshed myself;  When I try to access cybermega
in /var/lib/asterisk/static-http at port 8088 the asterisk debug shows:

[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session.  Top level
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/
index.html
[Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html]
has no handler
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open.
status_code:404

When using static-http you have to have /static at the front so the path would 
be:
/static/cyber/index.html


When I serve it from apache, the web ui appears, but never connects.

Using the firefox dev tools/console I see firefox can't establish a
connection the server at wss://:8089/ws

The asterisk debug log shows:

[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session.  Top level
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws
[Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no
handler
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open.
status_code:404

Suggestions?

Is there anything in the console at startup stating that stuff didn't load? The 
module which does websockets is res_http_websocket, and you can see if all that 
is needed is loaded using:

"module show like websocket"

  on the CLI.

That's a step closer!  The URI has to be 
/asterisk/static/cyber/index.html. I should make the asterisk part go 
away, but for now...
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Re: [asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 04/24/2018 09:08 AM, Matt Fredrickson wrote:

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell <bferr...@baywinds.org> wrote:

A while back (last year maybe?), there was a Digium blog post on setting up
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!


OK, I've gone back and refreshed myself;  When I try to access cybermega in 
/var/lib/asterisk/static-http at port 8088 the asterisk debug shows:

[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session.  Top level
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/index.html
[Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] has no 
handler
[Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. 
status_code:404

When I serve it from apache, the web ui appears, but never connects.

Using the firefox dev tools/console I see firefox can't establish a connection the 
server at wss://:8089/ws

The asterisk debug log shows:

[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session.  Top level
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws
[Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no handler
[Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. 
status_code:404

Suggestions?




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Re: [asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 04/24/2018 09:08 AM, Matt Fredrickson wrote:

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell <bferr...@baywinds.org> wrote:

A while back (last year maybe?), there was a Digium blog post on setting up
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?

Great question!  I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?

I'd be curious as to what difficulties you ran into.  We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support

You might try those two pages if you haven't yet.  If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.

Thanks!


Matt,

That is indeed the post.  I could get as far as the second web page screenshot 
and nothing past that login/connect screen... and no meaningful logs on the 
asterisk instance.

I tried serving the client via the Apache instance on the server (2.2 and 2.4) 
and from the asterisk built in http(s) server.

I'm basically a hobbyist and occasional contractor.   The paying job beckoned, 
so...

I'd REALLY like to get it working.  And for the record, I REALLY HATE pjsip.

I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux since '93 and telecom since 1980.  The config is so opaque, poorly documented and error prone I, to this 
day, use the legacy sip config wherever I can.  No one has ever been able to show me an advantage for it and it doesn't seem to use realtime configuration (even more of a 
drawback).  I much prefer realtime for my configuration on Asterisk;  Having configuration picked up from a DB is far preferable to reloading flat files.




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[asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

A while back (last year maybe?), there was a Digium blog post on setting up 
WebRTC.

I was never able to get that working.

I was working with Asterisk 15 on a RHEL derived distro and had no idea of 
where to go to shoot the failure.

Has anyone got a tutorial with trouble shooting?




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Re: [asterisk-users] [OT] Load testing with SIPp

[Bruce Ferrell]

On 03/06/2018 05:47 PM, Olivier wrote:

All mentioned boxes are VMWare VMs probably configured with default drivers and 
settings.
Unfortunately, I don't know which network equipements are used when those boxes 
communicate with each other.
So yes, maybe I banged into an unexpected limit because of this ignorance.

Anyway, from my testing, I was mostly surprised  by 2 things:

1- I consistently hit a 200 calls when I let run an Asterisk instance on the SIPp (calls passing locally from SIPp to a first Asterisk instance before hitting SUT): I thought 
that communications inside a given VM was "unlimited and cheap" and I was apparently wrong,

Disabling Asterisk in my first VM allowed me to hit a 500 limit at the price of 
loosing Asterisk flexibility to log RTCP stats, select exotic codec and so on.



What you've banged into is called "over subscription".  NOTHING is unlimited and 
"cheap"  There Ain't No Such Thing As A Free Lunch.

Virtualization works best when the guest VMs aren't 100% duty cycle... When one guest is idle, another can use the resources of the host.  When the guest VMs all present a 
significant load in any area, the host WILL bog down and so will all of the guest systems.


You might be better of not running this as a virtual app.

2- Increasing asterisk.conf's maxfiles value (from 1 to 4, for instance), allowed me to get more than 695 successfull call out of 700 but I could succeed to get, even 
once, 700 successfull calls, even when I tried with my current 40 maxfiles limit.


See the above... While you tweaked maxfiles on the guest, the host came came 
into play.


2018-03-06 23:35 GMT+01:00 Bruce Ferrell <bferr...@baywinds.org 
<mailto:bferr...@baywinds.org>>:


On 03/06/2018 01:58 PM, Olivier wrote:

Hello,

I'm running load testing sessions.
My System Under Test is an asterisk 13 with 16GB, configured with 
maxfiles set to 400 000.
This system is supposed do produce simple SIP trunking services without 
transcoding.


The box sending call to my System Under Test is anabled with SIPp.
I'm banging on a 700 concurrent calls/50 CAPS limit I would like to 
improve, if possible.

Tests are done with both signaling and media like this:

SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media

I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to 
SUT or SUT to echoing box) using iperf3 TCP testing though my target relies on 
UDP


My questions are:

1. Have you ever noticed a better scalability using UDP or TCP ?

2. Where do Retransmission I'm observing on SIPp console most probably 
come from ? Network issues ? My SIPp not beeing correctly tuned ? Lack of 
resources somewhere ?

3. Recommandations ? Suggestions ?

Best


I do network management for a living.

In your description, I see nothing to describe the network other than an 
observed 930Mb/s.

What is the network configuration;  What NIC(s), switches etc.

Treating these as effectively "unlimited" is a certain recipe for banging 
into unexpected limits.

Different NICs and switchs can and do provide differing levels of 
performance.




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Re: [asterisk-users] [OT] Load testing with SIPp

[Bruce Ferrell]

On 03/06/2018 01:58 PM, Olivier wrote:

Hello,

I'm running load testing sessions.
My System Under Test is an asterisk 13 with 16GB, configured with maxfiles set 
to 400 000.
This system is supposed do produce simple SIP trunking services without 
transcoding.


The box sending call to my System Under Test is anabled with SIPp.
I'm banging on a 700 concurrent calls/50 CAPS limit I would like to improve, if 
possible.

Tests are done with both signaling and media like this:

SIPp <---> SUT (asterisk 13) <---> Asterisk box echoing media

I checked bandwidth first and got 930 Mb/s on each leg (from SIPp to SUT or SUT 
to echoing box) using iperf3 TCP testing though my target relies on UDP


My questions are:

1. Have you ever noticed a better scalability using UDP or TCP ?

2. Where do Retransmission I'm observing on SIPp console most probably come 
from ? Network issues ? My SIPp not beeing correctly tuned ? Lack of resources 
somewhere ?

3. Recommandations ? Suggestions ?

Best



I do network management for a living.

In your description, I see nothing to describe the network other than an 
observed 930Mb/s.

What is the network configuration;  What NIC(s), switches etc.

Treating these as effectively "unlimited" is a certain recipe for banging into 
unexpected limits.

Different NICs and switchs can and do provide differing levels of performance.



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Re: [asterisk-users] Is this the future of telephony?

[Bruce Ferrell]

What might be good to consider is that "traditional" telcos actually use VOIP 
tech on the back end a lot.

SIP URIs seem to be a kewl and ubiquitous solution, but for the average joe, 
not so easy to setup and use.

And for the record, I consider wireless carriers as traditional telcos as well 
as wireline carriers.


On 06/15/2017 05:56 PM, Christopher van de Sande wrote:

Hey, so not particularly asterisk related, but I think the people on
this list are the right bunch to ask.

So does anyone here think the traditional telephone company will go
extinct, and voice communication will take place via email like (or
equal to) sip uri's?

I just setup an anonymous endpoint in pjsip.conf and a context that
forwards to $EXTEN and when I setup the correct SRV records, it seems
that any SIP client that's smart enough can just dial my SIP/email
address.  Is this what the future looks like?







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Re: [asterisk-users] CDR reporting solution

[Bruce Ferrell]

How about CDR to either MySQL or cdrlite and a quickie sql query? I could have 
added postgres, but I'm a DB bigot.  That would work too.

On 03/22/2017 01:46 PM, Motty Cruz wrote:


Hello, I am looking for CDR reporting solution? Any suggestions? I am using 
Asterisk 13.13.1

I would like a report on number of calls per extension.

Thanks,
Motty






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Re: [asterisk-users] failing to start asterisk on centos7

[Bruce Ferrell]

You didn't mention any of these and they may seem unimportant but they are:

Was the install from source or rpm?
Is this a new Centos install?

A brand new Centos install has selinux engaged.  selinux will block a 
source installed asterisk.  I think the asterisk RPMs have the necessary 
labeling to bypass selinux... I'm not sure of that. test for selinux with:


the response can be enabled, permissive or disabled

enabled means selinux is possibly the problem and the solution can be 
tested by doing:


setenforce 0

and then attempt to restart asterisk.  If it starts, it's a good 
indication of what is going on and I'll leave the exercise of either 
permenantly disabling selinux or correctly labeling the filesystem to 
allow selinux and asterisk to co-exist.


On 12/10/16 10:24 AM, christopher kamutumwa wrote:
ive installed asterisk but below is what am getting proces gets 
killed.please help


[root@localhost sounds]# asterisk -c
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer >
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' 
for details.
This is free software, with components licensed under the GNU General 
Public
License version 2 and other licenses; you are welcome to redistribute 
it under

certain conditions. Type 'core show license' for details.
=
[ Initializing Custom Configuration Options ]
  == Parsing '/etc/asterisk/extconfig.conf': Found
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Manager registered action DBGet
  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
PBX UUID: 36c0be35-766e-46c4-be1f-96479afa1e2b
  == Registered 'audio' codec 'g723' at sample rate '8000' with id '1'
  == Created cached format with name 'g723'
  == Registered 'audio' codec 'ulaw' at sample rate '8000' with id '2'
  == Created cached format with name 'ulaw'
  == Registered 'audio' codec 'alaw' at sample rate '8000' with id '3'
  == Created cached format with name 'alaw'
  == Registered 'audio' codec 'gsm' at sample rate '8000' with id '4'
  == Created cached format with name 'gsm'
  == Registered 'audio' codec 'g726' at sample rate '8000' with id '5'
  == Created cached format with name 'g726'
  == Registered 'audio' codec 'g726aal2' at sample rate '8000' with id '6'
  == Created cached format with name 'g726aal2'
  == Registered 'audio' codec 'adpcm' at sample rate '8000' with id '7'
  == Created cached format with name 'adpcm'
  == Registered 'audio' codec 'slin' at sample rate '8000' with id '8'
  == Created cached format with name 'slin'
  == Registered 'audio' codec 'slin' at sample rate '12000' with id '9'
  == Created cached format with name 'slin12'
  == Registered 'audio' codec 'slin' at sample rate '16000' with id '10'
  == Created cached format with name 'slin16'
  == Registered 'audio' codec 'slin' at sample rate '24000' with id '11'
  == Created cached format with name 'slin24'
  == Registered 'audio' codec 'slin' at sample rate '32000' with id '12'
  == Created cached format with name 'slin32'
  == Registered 'audio' codec 'slin' at sample rate '44100' with id '13'
  == Created cached format with name 'slin44'
  == Registered 'audio' codec 'slin' at sample rate '48000' with id '14'
  == Created cached format with name 'slin48'
  == Registered 'audio' codec 'slin' at sample rate '96000' with id '15'
  == Created cached format with name 'slin96'
  == Registered 'audio' codec 'slin' at sample rate '192000' with id '16'
  == Created cached format with name 'slin192'
  == Registered 'audio' codec 'lpc10' at sample rate '8000' with id '17'
  == Created cached format with name 'lpc10'
  == Registered 'audio' codec 'g729' at sample rate '8000' with id '18'
  == Created cached format with name 'g729'
  == Registered 'audio' codec 'speex' at sample rate '8000' with id '19'
  == Created cached format with name 'speex'
  == Registered 'audio' codec 'speex' at sample rate '16000' with id '20'
  == Created cached format with name 'speex16'
  == Registered 'audio' codec 'speex' at sample rate '32000' with id '21'
  == Created cached format with name 'speex32'
  == Registered 'audio' codec 'ilbc' at sample rate '8000' with id '22'
  == Created cached format with name 'ilbc'
  == Registered 'audio' codec 'g722' at sample rate '16000' with id '23'
  == Created cached format with name 'g722'
  == Registered 'audio' codec 'siren7' at sample rate '16000' with id '24'
  == Created cached format with name 'siren7'
  == Registered 'audio' codec 'siren14' at sample rate '32000' with id 
'25'

  == Created cached format with name 'siren14'
  == Registered 'audio' codec 'testlaw' at sample rate '8000' with id '26'
  == Created cached format with name 'testlaw'
  == Registered 'audio' codec 'g719' at sample rate '48000' with id '27'
  == Created cached format with 

Re: [asterisk-users] Call a subroutine via Originate?

[Bruce Ferrell]

use the macro construct and return from the macro

On 5/3/16 2:24 PM, John Kiniston wrote:

Howdy everyone,

I'm writing a little click to dial type tool and I've run into a snag 
where my Originate command needs to call a Sub routine to do a 
database lookup and some other stuff.


I can't seem to get the syntax right to call Gosub with Originate

Just testing with the command line I've been unable to make it work 
with any of these attempts:


originate PJSIP/johntest application Gosub sub-callout s,1

originate PJSIP/johntest application Gosub sub-callout(s,1)

originate PJSIP/johntest application Gosub (sub-callout,s,1)

What Syntax should I be using?

And if it helps I'll be calling this via AMI over https.

Thanks!

--
A human being should be able to change a diaper, plan an invasion, 
butcher a hog, conn a ship, design a building, write a sonnet, balance 
accounts, build a wall, set a bone, comfort the dying, take orders, 
give orders, cooperate, act alone, solve equations, analyze a new 
problem, pitch manure, program a computer, cook a tasty meal, fight 
efficiently, die gallantly. Specialization is for insects.

---Heinlein




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Re: [asterisk-users] Help with CDR-Stats

[Bruce Ferrell]

billing is sending invoices for calls to customers.

reporting is overall statistics on the aggregate of your calls... 
Average call hold time, common (or uncommon destinations) etc.  If you 
see a destination that suddenly has a lot of calls with hold time below 
normal, there may be a call quality problem.


TPC (the phone company) has used statistical troubleshooting techniques 
for decades to keep quality up so customers don't have to complain, not 
to mention for sizing.




On 12/16/15 8:23 AM, Vitor Mazuco wrote:


Humm whats is the diferent?

Em 16/12/2015 14:19, "Annus Fictus" > escreveu:


CDR-STATS is for reporting.

A2Billing is for billing...

Regards

El 16/12/2015 a las 11:15, Vitor Mazuco escribió:

Hi everyone!

I'm trying to install CDR-Stats (cdr-stats.org
), but it very difficult.

Is there others optins for billing?

Thanks



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Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

[Bruce Ferrell]

just got back to my mail.

What I'd do is go into /etc/asterisk and grep for OUT_3_SUFFIX in all the files

once the file with that variable is located, we can figure out why it's adding 
it



On 08/17/2015 11:26 PM, David Cunningham wrote:
 Yes indeed.

 Do you have the dialplan, eg from /etc/asterisk/extensions.conf?

 Something is getting this OUT_3_SUFFIX variable and including it in a Dial to 
 172.22.4.12.


 On 18 August 2015 at 16:21, Brendan Ord b...@staff.onthenet.com.au 
 mailto:b...@staff.onthenet.com.au wrote:

 Starting to make sense when I saw this line:

  

 [2015-08-18 15:01:33] DEBUG[19366][C-1cfc]: pbx.c:3785 
 ast_str_retrieve_variable: Result of 'OUT_3_SUFFIX' is '@CUBE'

  

 But I can’t find where this is in configuration ..

  

 Brendan Ord
 OntheNet - Network Engineer
 P 07 5553 9222
 F 07 5593 3557
 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map 
 https://goo.gl/maps/p25WF)
 www.OntheNet.com.au http://www.onthenet.com.au/

  

 *From:*asterisk-users-boun...@lists.digium.com 
 mailto:asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com
 mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Brendan 
 Ord
 *Sent:* Tuesday, 18 August 2015 3:44 PM


 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk appending 
 @string to dialled number

  

 David,

  

 I should also note;

  

 246 is my extension, it has IP 172.22.3.238.

  

 172.22.4.8 is the PBX, and 172.22.4.12 is the 2800 gateway.

  

 The trunk is called ‘testing’ at the moment.  The route that selects this 
 trunk uses a 9 prefix.

  

 This system is in semi-production, so there might be fluff in the log 
 from other active calls.

  

 Brendan Ord
 OntheNet - Network Engineer
 P 07 5553 9222
 F 07 5593 3557
 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map 
 https://goo.gl/maps/p25WF)
 www.OntheNet.com.au http://www.onthenet.com.au/

  

 *From:*asterisk-users-boun...@lists.digium.com 
 mailto:asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com
 mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David 
 Cunningham
 *Sent:* Tuesday, 18 August 2015 2:39 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk 13 chan_sip trunk appending 
 @string to dialled number

  

 Hi Brendan,

 Can you attach an Asterisk log with sip set debug on, core set verbose 
 9 and core set debug 9?

  

 On 18 August 2015 at 10:33, Brendan Ord b...@staff.onthenet.com.au 
 mailto:b...@staff.onthenet.com.au wrote:

 Hello,

  

 I’m having what seems like a weird issue connecting Asterisk 13 (FreePBX 
 12) to a Cisco 2811 router via a chan_sip trunk.  Whenever I try dialling out 
 via this trunk,
 something appends ‘@CUBE’ onto the end of the dialled number, as per the 
 following examples;

  

 Asterisk log;

 app_dial.c: Called SIP/test/0429123456@CUBE

 chan_sip.c: Got SIP response 500 Internal Server Error back from 
 172.22.4.12:5060 http://172.22.4.12:5060

  

 In the SIP SDP;

 INVITE sip:0429920437%40CUBE@172.22.4.12 
 mailto:sip%3A0429920437%2540CUBE@172.22.4.12 SIP/2.0.

 To: sip:0429920437%40CUBE@172.22.4.12 
 mailto:sip%3A0429920437%2540CUBE@172.22.4.12.

  

 As you can see, the @CUBE carries over into the SIP URI as %40CUBE.  The 
 FPBX trunk name and outbound route were called CUBE (afaik, purely 
 descriptive) but I changed them to
 something different and the @CUBE persisted.  I’m really not sure where 
 this is coming from, and why.

  

 Here is my trunk configuration;

  

 PEER

 type=friend

 qualify=yes

 nat=no

 insecure=port,invite

 host=172.22.4.12

 dtmfmode=rfc2833

 context=from-trunk

 allow=ulaw

 disallow=all

  

 USER

 type=friend

 qualify=yes

 nat=no

 host=172.22.4.12

 dtmfmode=rfc2833

 allow=ulaw

 disallow=all

 canreinvite=no

  

 Thanks for any help J

  

 Brendan Ord
 OntheNet - Network Engineer
 P 07 5553 9222
 F 07 5593 3557
 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map 
 https://goo.gl/maps/p25WF)
 www.OntheNet.com.au http://www.onthenet.com.au/

  


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Re: [asterisk-users] Shared RealTime Database

[Bruce Ferrell]

yes, the sip_buddies (or equal) has a field that says which server handled the 
registration

On 08/17/2015 07:58 AM, Mehdi Shirazi wrote:
 Hi
 If we have a shared RealTime database for sip registration of multiple
 Asterisk servers, is there a way to realize which Asterisk server registered 
 sip phones ?

 Regards
 M.Shirazi 




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Re: [asterisk-users] Asterisk 13 chan_sip trunk appending @string to dialled number

[Bruce Ferrell]

Brenden,

check the context, from-trunk, in the dialplan.  Thtat's where this is 
being added




On 8/17/15 5:33 PM, Brendan Ord wrote:


Hello,

I’m having what seems like a weird issue connecting Asterisk 13 
(FreePBX 12) to a Cisco 2811 router via a chan_sip trunk. Whenever I 
try dialling out via this trunk, something appends ‘@CUBE’ onto the 
end of the dialled number, as per the following examples;


Asterisk log;

app_dial.c: Called SIP/test/0429123456@CUBE

chan_sip.c: Got SIP response 500 Internal Server Error back from 
172.22.4.12:5060


In the SIP SDP;

INVITE sip:0429920437%40CUBE@172.22.4.12 SIP/2.0.

To: sip:0429920437%40CUBE@172.22.4.12.

As you can see, the @CUBE carries over into the SIP URI as %40CUBE.  
The FPBX trunk name and outbound route were called CUBE (afaik, purely 
descriptive) but I changed them to something different and the @CUBE 
persisted.  I’m really not sure where this is coming from, and why.


Here is my trunk configuration;

PEER

type=friend

qualify=yes

nat=no

insecure=port,invite

host=172.22.4.12

dtmfmode=rfc2833

context=from-trunk

allow=ulaw

disallow=all

USER

type=friend

qualify=yes

nat=no

host=172.22.4.12

dtmfmode=rfc2833

allow=ulaw

disallow=all

canreinvite=no

Thanks for any help J

Brendan Ord
OntheNet - Network Engineer
P 07 5553 9222
F 07 5593 3557
Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map 
https://goo.gl/maps/p25WF)

www.OntheNet.com.au http://www.onthenet.com.au/





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Re: [asterisk-users] Anonymous SIP calls

[Bruce Ferrell]

James,

I'm a systems and telecom professional with experience going back more than 
thirty years, to the days of teletype, current loop, POTS (2600hz signalling 
anyone?) and echo
cancellation via analog level control and hybrid balance.

Your read of the intent of the VOIP/SIP design correctly.  The intent WAS to 
make making connections between endpoints as easy as using a browser.

Unfortunately, setting up ALL of the infrastructure, not JUST the 
registration/switching points (Asterisk/Kamailiao/Freeswitch), can be quite 
daunting... In general, simple DNS is
beyond most and the necessary specialized (and they aren't That SPECIAL)  SRV 
records make most systems admins run for the hills these days. 

When we see a statement regarding consideration of allowing anonymous calls, we 
seeing someone who is (rightly) concerned about fraudulent use of an expensive 
resource... PSTN
interconnect.  In the intended vision, that would be a don't care scenario, 
because the PSTN interconnect wouldn't exist, but it does and it's billed by 
it's use making it expensive.

In theory, E164 would have take up closer to that ideal.  Asterisk has hooks 
and connections to use it and it's own, competing directory mechanism, DUNDi.  
Let's make special note
of a word I used in that last sentence... Competing.  Is DUNDi better?  I don't 
know and I'm fairly certain I just touched off a debate on the topic.  But I do 
know that when
things start competing/contending, people do a few things:

1.) They take sides and fragment things
2.) The sit on the sidelines and wait for things to settle out

In my experience, this has a tendency to bring things to a halt.

Add to this, most of this tech is really, really only useful to businesses.  I 
give my skills to people who need it (Family, friends my old gray haired 
mother-in-law).  Businesses
are in the business of making money and if they want the use of my skills, they 
get to pay me.  No one I know will perform this type of thing for free for a 
business and we all
compete for the limited pool of resource that business is willing to offer.  
What I have to offer is the tricks of the trade I've garnered over a lifetime 
career.  There was a
time when systems admins freely swapped these tips, tricks and techniques (for 
the best example see the old Novell Users FAQ).  As I mentioned before, we who 
know how to install
and maintain VOIP systems are now competing and the dollars come hard, so there 
seems (at least in the areana of VOIP) less willingness to do this.  Oddly, 
VOIP seems to be more
cut throat that any other sector of IT.

Just my experience and I'm sticking to it... and wishing it weren't so and that 
unicorns really existed.



On 03/27/2015 01:03 PM, James B. Byrne wrote:
 On Thu, March 26, 2015 22:29, Michelle Dupuis wrote:
 You have to consider whether you really want anonymous calls, or you
 just want to enable SIP calls from trusted companies/partners.  The
 latter means setting up routes to these companies and (ideally)
 registration between peers.

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

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

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

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

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

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

 I somewhat understand the process of getting devices to register and
 authenticate to obtain access to our outgoing routes.   What is it
 about incoming SIP calls destined to our internal users that make
 those calls so dangerous?  Why cannot incoming anonymous SIP calls not
 be treated exactly as incoming PSTN calls (other than 

Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

[Bruce Ferrell]

On 11/14/2014 02:57 PM, Kevin Larsen wrote:
   Hi,
  
   Change my Dynastar E1 gateway to Cisco with E1 module, but can't make
   easiest dialplan. All my routing i made on asterisk, so i need that cisco
   all calls from E1 send via sip to Asterisk and all calls came from 
   Asterisk
   by sip send to E1. From E1 to Asterisk already work, but can't understand
   how send all from Asterisk SIP to E1 ?
  
   Can you help ?
 
  If you are taking SIP calls into Asterisk and want to send them out
  E1, you need an Asterisk-compatible E1 board, such as:

 I think, though I am not for sure, that he is asking how to route a call from 
 Asterisk to the Cisco device that is acting as the gateway (i.e. the E1 is 
 connected to the Cisco
 and will only be speaking SIP to Asterisk).

 We had a similar setup before we replaced the Cisco with Digium Gateways. 
 Basically we just set up a peer in asterisk with the IP address of the Cisco 
 and routed external calls
 to it. The Cisco then actually placed the calls to the PSTN. Pretty painless, 
 though sometimes a bit confusing to remember that if there were issues, you 
 might have to check
 dialing rules in both Asterisk and in the Cisco.

Yep,  I concur, that's what he's asking for. The syntax for Cisco dial-peers 
isn't a whole lot of fun, but not horrible either and once you know to call it 
a dial-peer, the
documentation is easy to google.


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[asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

I was just told that realtime was no longer in asterisk 12, but I find 
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

Yeah, and I didn't find anything there.  I was looking for something a little more 
concrete that it should be...



On 04/17/2014 01:16 PM, Eric Wieling wrote:

All significant changes should be listed in the UPGRADE*.txt included in the 
Asterisk source code.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Ferrell
Sent: Thursday, April 17, 2014 4:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

I was just told that realtime was no longer in asterisk 12, but I find 
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


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Re: [asterisk-users] Realtime in asterisk 12 removed/deprecated?

[Bruce Ferrell]

On 04/17/2014 01:34 PM, Joshua Colp wrote:

Bruce Ferrell wrote:

I was just told that realtime was no longer in asterisk 12, but I find
enhancements in 12.2-rc2 and no sign in the wiki that this is true.

Can someone comment?


Realtime has not been removed or deprecated. A new model for newly written modules has been created, but nothing existing has been migrated to it or even will be (it's a 
fundamentally shift).



Thanks Joshua!

Always good to have a definitive statements

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[asterisk-users] Unable to build DAHDI

[Bruce Ferrell]

Hi all,

I'm attempting to install dahdi-linux-complete-2.6.1+2.6.1 on an OpenSUSE 
system running kernel 3.7.1-17

The build is failing like this:

ct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tone_detection.c oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsi_cnct.c 
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c oct612x/apilib/bt/octapi_bt0.c oct612x/apilib/largmath/octapi_largmath.c oct612x/apilib/llman/octapi_llman.c

oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c: In function 
‘Oct6100ApiBridgeEventRemove’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_conf_bridge.c:3870:47: 
note: each undeclared identifier is reported only once for each function it 
appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100EventGetToneDef’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:89:32: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:89:32: note: each 
undeclared identifier is reported only once for each function it appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100BufferPlayoutTransferEvents’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:1116:126: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c: In function 
‘Oct6100BufferPlayoutCheckForSpecificEvent’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_events.c:1287:29: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c: In 
function ‘Oct6100ApiStrStr’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c:391:38: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_miscellaneous.c:391:38: 
note: each undeclared identifier is reported only once for each function it 
appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c: In function 
‘Oct6100BufferPlayoutStopDef’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:530:43: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:530:43: note: 
each undeclared identifier is reported only once for each function it appears in
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c: In function 
‘Oct6100ApiInvalidateChanPlayoutStructs’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_playout_buf.c:2844:51: 
error: ‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c: In function 
‘Oct6100ApiReserveTsst’:
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c:401:29: error: 
‘NULL’ undeclared (first use in this function)
oct612x/octdeviceapi/oct6100api/oct6100_api/oct6100_tsst.c:401:29: note: each 
undeclared identifier is reported only once for each function it appears in
oct612x/apilib/bt/octapi_bt0.c: In function ‘OctApiBt0Init’:
oct612x/apilib/bt/octapi_bt0.c:63:12: error: ‘NULL’ undeclared (first use in 
this function)
oct612x/apilib/bt/octapi_bt0.c:63:12: note: each undeclared identifier is 
reported only once for each function it appears in
oct612x/apilib/llman/octapi_llman.c: In function ‘OctapiLlmAllocInit’:
oct612x/apilib/llman/octapi_llman.c:88:12: error: ‘NULL’ undeclared (first use 
in this function)
oct612x/apilib/llman/octapi_llman.c:88:12: note: each undeclared identifier is 
reported only once for each function it appears in
make[3]: *** [oct6100_adpcm_chan.o] Error 1
make[3]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools/xpp'
make[2]: *** [utils-subdirs] Error 2
make[2]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools'
make[1]: *** [all] Error 2
make[1]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.6.1+2.6.1/tools'
make: *** [all] Error 2


Any suggestions are welcome



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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

Jonas,

May I suggest that you present us your sip.conf entry for this peer, properly 
redacted, of course.  That might help more.  What I do for gateways at known 
addresses is to put an
entry like this into the sip.conf entry:


[peer]
type=peer
defaultip=192.168.40.123
insecure=invite,port
context=some_context





On 11/22/2011 06:40 AM, Jonas Kellens wrote:
 Hello list,

 this is the communication between an Aastra 5000 PBX and Asterisk, where the 
 Aastra makes a call to Asterisk. For some reason, Asterisk responds with 
 401-Unauthorized and I don't
 know why.

 Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with 
 this Aastra.


 A1.A1.A1.A1 = IP-address Asterisk PBX
 AS.AS.AS.AS = IP-address Aastra PBX

 Aastra PBX makes a call to the number 3221112233...

 This is all the sip debug trace gathered with asterisk :


 --- SIP read from UDP:AS.AS.AS.AS:61490 ---
 INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
 From: sip:sippeerusern...@sip.domain.tld:5060;tag=310158BD
 To: sip:3221112233@A1.A1.A1.A1:5060
 Call-ID: 0201CEFEA742
 CSeq: 1 INVITE
 Contact: sip:sippeerusern...@as.as.as.as:61490
 Proxy-Authorization: Digest username=SIPPEERusername, realm=domain.tld, 
 nonce=67105ac4, uri=sip:3221112233@A1.A1.A1.A1:5060, response=60be856773
 f86450fc9ddbaf7a568505, algorithm=MD5
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
 Max-Forwards: 70
 Privacy: none
 P-Asserted-Identity: sip:3224445...@sip.domain.tld
 User-Agent: A5000 R52-H2C0205
 P-Behind-Gsi: 192.168.6.1
 Content-Type: application/sdp
 Content-Length:195

 v=0
 o=- 0 0 IN IP4 sip.domain.tld
 s=-
 i=(o=IN IP4 10.1.2.35)
 c=IN IP4 AS.AS.AS.AS
 t=0 0
 m=audio 62654 RTP/AVP 8 0
 a=rtcp:65115
 a=rtpmap:8 PCMA/8000
 a=rtpmap:0 PCMU/8000
 a=ptime:20

 -

 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (16 headers 
 11 lines) ---
 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using SIP 
 RTP TOS bits 184
 [Nov 18 15:14:35] VERBOSE[2255] netsock.c: [Nov 18 15:14:35]   == Using SIP 
 RTP CoS mark 5
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Sending to 
 AS.AS.AS.AS : 61490 (no NAT)
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Using INVITE 
 request as basis request - 0201CEFEA742
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Found peer 
 'SIPPEERusername' for 'SIPPEERusername' from AS.AS.AS.AS:61490
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
 --- Reliably Transmitting (NAT) to AS.AS.AS.AS:61490 ---
 SIP/2.0 401 Unauthorized
 Via: SIP/2.0/UDP 
 AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;received=AS.AS.AS.AS;rport=61490
 From: sip:sippeerusern...@sip.domain.tld:5060;tag=310158BD
 To: sip:3221112233@A1.A1.A1.A1:5060;tag=as68f71fe5
 Call-ID: 0201CEFEA742
 CSeq: 1 INVITE
 Server: Asterisk PBX 1.6.2.20
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 WWW-Authenticate: Digest algorithm=MD5, realm=domain.tld, nonce=46ef24d9
 Content-Length: 0

 
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] Scheduling 
 destruction of SIP dialog '0201CEFEA742' in 32000 ms (Method: INVITE)
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
 --- SIP read from UDP:AS.AS.AS.AS:61490 ---
 ACK sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK15388160301891243008;rport
 From: sip:sippeerusern...@sip.domain.tld:5060;tag=310158BD
 To: sip:3221112233@A1.A1.A1.A1:5060;tag=as68f71fe5
 Call-ID: 0201CEFEA742
 CSeq: 1 ACK
 Contact: sip:sippeerusern...@as.as.as.as:61490
 Max-Forwards: 70
 User-Agent: A5000 R52-H2C0205
 P-Behind-Gsi: 192.168.6.1
 Content-Length: 0


 -
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35] --- (11 headers 
 0 lines) ---
 [Nov 18 15:14:35] VERBOSE[2255] chan_sip.c: [Nov 18 15:14:35]
 --- SIP read from UDP:AS.AS.AS.AS:61490 ---
 INVITE sip:3221112233@A1.A1.A1.A1:5060 SIP/2.0
 Via: SIP/2.0/UDP AS.AS.AS.AS:61490;branch=z9hG4bK6996345481960200906;rport
 From: sip:sippeerusern...@sip.domain.tld:5060;tag=33015DBD
 To: sip:3221112233@A1.A1.A1.A1:5060
 Call-ID: 0201CCFEA242
 CSeq: 1 INVITE
 Contact: sip:sippeerusern...@as.as.as.as:61490
 Proxy-Authorization: Digest username=SIPPEERusername, realm=domain.tld, 
 nonce=46ef24d9, uri=sip:3221112233@A1.A1.A1.A1:5060, 
 response=14ecbfc7df24b49926151284c123ea11,
 algorithm=MD5
 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE
 Max-Forwards: 70
 Privacy: none
 P-Asserted-Identity: sip:3224445...@sip.domain.tld
 User-Agent: A5000 R52-H2C0205
 P-Behind-Gsi: 192.168.6.1
 Content-Type: application/sdp
 Content-Length:195

 v=0
 o=- 0 0 IN IP4 sip.domain.tld
 s=-
 i=(o=IN IP4 10.1.2.35)
 c=IN IP4 AS.AS.AS.AS
 t=0 0
 m=audio 62654 RTP/AVP 8 0
 a=rtcp:65115
 a=rtpmap:8 PCMA/8000
 a=rtpmap:0 PCMU/8000
 

Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

On 11/22/2011 07:29 AM, Jonas Kellens wrote:
 On 11/22/2011 04:25 PM, Bruce Ferrell wrote:

 Jonas,

 May I suggest that you present us your sip.conf entry for this peer, 
 properly redacted, of course.  That might help more.  What I do for 
 gateways at known addresses is to put
 an entry like this into the sip.conf entry:


 [peer]
 type=peer
 defaultip=192.168.40.123
 insecure=invite,port
 context=some_context



 This is the peer definition in sip.conf :

 [SIPPEERusername]
 type=friend
 host=dynamic
 defaultuser=SIPPEERusername
 secret=guessthis
 context=from-PEERTRUNK
 nat=yes
 dtmfmode=rfc2833
 canreinvite=no
 disallow=all
 allow=alaw
 allow=gsm


 Hope you can help me out with this extra information.


 Kind regards,

 Jonas.
From what I see in your entry, you are requiring registration from the peer.  
The next thing i would check is to see if the registration has succeeded.  If 
it doesn't succeed, you
will see the results you presented.  I see you have the peer set as a dynamic 
host, and if the IP address of the device does in fact change then registration 
is appropriate.
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Re: [asterisk-users] Asterisk refuses INVITE (401) and I don't know why

[Bruce Ferrell]

Jonas,

I did see the traces and I agree they accurately show what you described.  What 
traces never do is say WHY it's happening.  What I suggest is that you go into 
the cli as root with:

asterisk -r

issue the following commands:

 core set debug 9
 core set verbose 5
 quit

and then monitor the logs found in /var/log/asterisk (that's where they are 
usually anyway)

you will most likely find the cause in either the debug, full or messages file, 
again assuming you have those files enabled.  If you don't have them enabled, 
please contact me,
either directly or via the list and I'll help you get them enabled and assist 
you in interpreting them.

I suggest that you NOT post them to the list as they could be quite large and a 
risk of accidentally revealing something you might not want to reveal.


On 11/22/2011 08:51 AM, Jonas Kellens wrote:
 On 11/22/2011 05:42 PM, Alex Vishnev wrote:
 I doubt it. Unknown headers should be ignored by UAS. is it possible to post 
 the trace?
 On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:

 What trace do you need ? Have you read my original post ? Asterisk SIP debug 
 trace is posted in my original post.


 Kind regards,
 Jonas.

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Re: [asterisk-users] Calling an independent gateway from asterisk

[Bruce Ferrell]

Amar,

In general, gateways don't register.  They are simply defined as a peer and 
calls are routed to them in the dialplan.  When I do this I usually use the 
local channel to get to the
dialing contexts.

Get in touch if you need a more detailed example

Bruce Ferrell

On 11/14/2011 10:01 PM, Amar Akshat wrote:
 Hello,
 I have a testing scenario at hand. I want to make a call from Asterisk
 CLI or AMI to an external network gateway. Is this possible.
 Let me explain the use case.

 Asterisk server (say 192.168.5.10) has few registered endpoints or softphone.
 Now an external gateway (say my-example.com or XXX.XXX.XXX.XXX:5060),
 listening for SIP invites, but this gateway is not registered with
 Asterisk,

 can I send out SIP invites (call) to this external gateway, without
 having to register on Asterisk.



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Re: [asterisk-users] Can we use MySQL native connector for ARA?

[Bruce Ferrell]

On 10/20/2011 05:59 AM, JR Richardson wrote:
 Hello Everyone,

 The documentation suggests using unixodbc for asterisk realtime. Is
 there any way
 we can just use native database clients such as libmysqlclient from
 MySQL? The native
 clients tend to be more up-to-date.

 Thanks in Advance,

 Nick.
 I've used the MySQL addon for years with great success, initiated the
 project to support read/write to separate databases for asterisk clustering.
 You will get more functionality for complex queries using the odbc
 connector, but for basic ARA applications, the MySQL addon works fine and
 I've never had a problem with stability.  I also use the cdr_mysql as well.

 I wrote a couple of papers on asterisk_clustering_with_mysql_replication.
 They are a bit dated but still relevant.  I'll send over if you like.

 Good luck.

 JR

Hey JR!

I've used the native MySQL connector too with great results.  Maybe you could 
post a link to the paper(s) on voip-info.org for posterity.  I know I'd love to 
seem them too

Bruce Ferrell

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Bruce Ferrell]

One thing I do is to use the mysql command to run simple queries and the
Goto a context with the information... simple and clean

On 09/25/2011 10:33 PM, Sam Govind wrote:
 Hmmm..interesting..I haven't came across anything like this so
 far..How about making a new table for the insertion of a new call
 data..and trigger some script to activate AMI/Call file according to
 new call data.

 http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10

 On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis sym...@gmail.com
 mailto:sym...@gmail.com wrote:

 Hello Everyone,

 I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
 wondering if it Is possible to have Asterisk make a calls based on a
 record inserted in a table realtime? If I have to develop
 something using AGI
 or AMI, I can do this  with a little direction?

 Thanks in Advance,

 Nick

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[asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

I know over time SIP OPTIONS message handling has changed and I've seen 
some write ups that seem to indicate that an s extension in the default 
context is needed now to get them to work.


It's probably my error in any case.

So, what am I doing wrong or what do I need to do to get the sip ping to 
work?


Bruce Ferrell

Just for fun, I created a sip peer called ping at a fixed address with a 
default context of ping:


sip.conf
===
[ping]
type=peer
host=XX.XX.XX.XX
defaultip=XX.XX.XX.XX
context=ping

extensions.conf

[ping]
exten = s,1,NoOp()


I am sending the following:


OPTIONS sip:YY.YY.YY.YY SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
From: sip:p...@xx.xx.xx.xx
To: sip:YY.YY.YY.YY
Contact: sip:p...@xx.xx.xx.xx
Call-ID: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
CSeq: 102 OPTIONS
User-Agent: sip_ping.pl
Date: Mon, 19 September 2011 08:57:47 PDT
Allow: ACK, CANCEL
Content-Length: 0


And the asterisk instance is logging the following with core set debug 9:

--- SIP read from UDP:XX.XX.XX.XX:6051 ---
OPTIONS sip:YY.YY.YY.YY SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
From: sip:p...@xx.xx.xx.xx
To: sip:YY.YY.YY.YY
Contact: sip:p...@xx.xx.xx.xx
Call-ID: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
CSeq: 102 OPTIONS
User-Agent: sip_ping.pl
Date: Mon, 19 September 2011 08:57:47 PDT
Allow: ACK, CANCEL
Content-Length: 0
-
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  0 [ 32]: OPTIONS 
sip:YY.YY.YY.YY SIP/2.0
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  1 [ 59]: Via: 
SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  2 [ 29]: From: 
sip:p...@xx.xx.xx.xx
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  3 [ 22]: To: 
sip:YY.YY.YY.YY
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  4 [ 32]: Contact: 
sip:p...@xx.xx.xx.xx
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  5 [ 54]: Call-ID: 
20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  6 [ 17]: CSeq: 
102 OPTIONS
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  7 [ 23]: 
User-Agent: sip_ping.pl
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  8 [ 41]: Date: 
Mon, 19 September 2011 08:57:47 PDT
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  9 [ 18]: Allow: 
ACK, CANCEL
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header 10 [ 18]: 
Content-Length: 0

[2011-09-19 08:57:47] VERBOSE[2979] chan_sip.c: --- (11 headers 0 lines) ---
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: = Looking for  Call ID: 
20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx (Checking From) --From tag  
--To-tag
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: OPTIONS request has no 
from tag, dropping callid: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx 
from: sip:p...@xx.xx.xx.xx
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: Invalid SIP message - 
rejected , no callid, len 355


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Re: [asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

On 09/19/2011 09:33 AM, Alex Balashov wrote:

Every request needs a From tag.



Uh... OK.  Isn't this a From tag:

From: sip:p...@xx.xx.xx.xx

Line three of what I send?


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Re: [asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

Thank you Alex.  That was useful information.   The sip_ping.pl program 
I was using used to work without the tag.  It seems asterisk now demands it.


Bruce Ferrell

On 09/19/2011 10:18 AM, Alex Balashov wrote:

On 09/19/2011 01:16 PM, Alex Vishnev wrote:


no, you need a tag i.e From: sip:p...@xx.xx.xx.xx;tag=xxx, where
xx is a unique identifier

see the definition of SIP Dialog

Dialog: A dialog is a peer-to-peer SIP relationship between two
  UAs that persists for some time.  A dialog is established by
  SIP messages, such as a 2xx response to an INVITE request.  A
  dialog is identified by a call identifier, local tag, and a
  remote tag.  A dialog was formerly known as a call leg in RFC
  2543.


OPTIONS requests don't create a dialog, just a transaction.




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[asterisk-users] Dahdi does not build against Kernel 3.0.0

[Bruce Ferrell]

Errors follow:

make
make -C linux all
make[1]: Entering directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux'

make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware'
make -C /lib/modules/3.0.0-39-desktop/build 
SUBDIRS=/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi 
DAHDI_INCLUDE=/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/include 
DAHDI_MODULES_EXTRA=  HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m

make[2]: Entering directory `/usr/src/linux-3.0.0-39-obj/i386/desktop'
make -C ../../../linux-3.0.0-39 
O=/usr/src/linux-3.0.0-39-obj/i386/desktop/. modules
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.o
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2: 
warning: #warning No CONFIG_BKL is an experimental configuration.
In file included from 
/usr/src/linux-3.0.0-39/arch/x86/include/asm/uaccess.h:570:0,

from /usr/src/linux-3.0.0-39/include/linux/poll.h:14,
from 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/include/dahdi/kernel.h:58,
from 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:68:

In function ‘copy_from_user’,
inlined from ‘dahdi_chan_write’ at 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:2301:23:
/usr/src/linux-3.0.0-39/arch/x86/include/asm/uaccess_32.h:211:26: 
warning: call to ‘copy_from_user_overflow’ declared with attribute 
warning: copy_from_user() buffer size is not provably correct
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2: 
warning: #warning No CONFIG_BKL is an experimental configuration.
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o
SHIPPED 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
LD [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_loc.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_eth.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_ethmf.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.c:49:31: 
error: ‘SPIN_LOCK_UNLOCKED’ undeclared here (not in a function)
make[5]: *** 
[/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o] 
Error 1
make[4]: *** 
[_module_/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi] 
Error 2

make[3]: *** [sub-make] Error 2
make[2]: *** [all] Error 2
make[2]: Leaving directory `/usr/src/linux-3.0.0-39-obj/i386/desktop'
make[1]: *** [modules] Error 2
make[1]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux'

make: *** [all] Error 2


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Re: [asterisk-users] Dahdi does not build against Kernel 3.0.0

[Bruce Ferrell]

Correction; the released DAHDI doesn't build svn does. Nothing to see 
here... Move along



On 08/06/2011 07:49 PM, Bruce Ferrell wrote:

Errors follow:

make
make -C linux all
make[1]: Entering directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux'

make -C drivers/dahdi/firmware firmware-loaders
make[2]: Entering directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware'
make[2]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/firmware'
make -C /lib/modules/3.0.0-39-desktop/build 
SUBDIRS=/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi 
DAHDI_INCLUDE=/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/include 
DAHDI_MODULES_EXTRA=  HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m

make[2]: Entering directory `/usr/src/linux-3.0.0-39-obj/i386/desktop'
make -C ../../../linux-3.0.0-39 
O=/usr/src/linux-3.0.0-39-obj/i386/desktop/. modules
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.o
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2: 
warning: #warning No CONFIG_BKL is an experimental configuration.
In file included from 
/usr/src/linux-3.0.0-39/arch/x86/include/asm/uaccess.h:570:0,

from /usr/src/linux-3.0.0-39/include/linux/poll.h:14,
from 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/include/dahdi/kernel.h:58,
from 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:68:

In function ‘copy_from_user’,
inlined from ‘dahdi_chan_write’ at 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:2301:23:
/usr/src/linux-3.0.0-39/arch/x86/include/asm/uaccess_32.h:211:26: 
warning: call to ‘copy_from_user_overflow’ declared with attribute 
warning: copy_from_user() buffer size is not provably correct
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi-base.c:95:2: 
warning: #warning No CONFIG_BKL is an experimental configuration.
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/dahdi_vpmadt032_loader.o
SHIPPED 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o
LD [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_loc.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_eth.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_dynamic_ethmf.o
CC [M] 
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o
/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.c:49:31: 
error: ‘SPIN_LOCK_UNLOCKED’ undeclared here (not in a function)
make[5]: *** 
[/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi/dahdi_transcode.o] 
Error 1
make[4]: *** 
[_module_/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux/drivers/dahdi] 
Error 2

make[3]: *** [sub-make] Error 2
make[2]: *** [all] Error 2
make[2]: Leaving directory `/usr/src/linux-3.0.0-39-obj/i386/desktop'
make[1]: *** [modules] Error 2
make[1]: Leaving directory 
`/usr/local/src/asterisk/dahdi-linux-complete-2.4.1.2+2.4.1/linux'

make: *** [all] Error 2


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[asterisk-users] Tutorial on the Asterisk Manager Interface

[Bruce Ferrell]

I've used the manager interface to make calls successfully, now I'd like 
a  look at some of  he other ways it can be used.


I've seen references to its use to perform call  cut off and rate CDRs.

Is anyone aware of a reference or tutorial I could look at?

Bruce Ferrell

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[asterisk-users] Realtime and priority labels

[Bruce Ferrell]

In the following example

exten = _1NXXNXX,1,Set(GROUP(outbound)=myprovider)
exten = _1NXXNXX,n,Set(COUNT=${GROUP_COUNT(myprovider@outbound)})
exten = _1NXXNXX,n,NoOp(There are ${COUNT} calls for myprovider)
exten = _1NXXNXX,n,GotoIf($[ ${COUNT}  2 ]?denied : continue)
exten = _1NXXNXX,n(denied),NoOp(There are too many calls up)
exten = _1NXXNXX,n,Hangup()
exten = _1NXXNXX,n(continue),GoSub(callmyprovider,${EXTEN},1)

instead of sequentially numbering the priorities, the n construct is 
used.  I find that when I attempt this in the realtime extensions table 
only, the first priority step is recognized.  If I sequentially number 
the priorities and add a label, the step is no longer recognized.


Is this behavior by design or an error?

Thanks in advance

Bruce



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Re: [asterisk-users] Prepaid Billing other than A2Billing

[Bruce Ferrell]

Nice promises, but no code :(

On 03/05/2011 10:34 PM, Pezhman Lali wrote:
 I think a2billing is the best billing opensource system, but try
 astbill, new url http://astbss.org/

 but if you want to setup a large system select enterprise system,
 these systems are useful for small and med networks.
 best

 On Sat, Mar 5, 2011 at 8:56 PM, bilal ghayyad bilmar...@yahoo.com
 mailto:bilmar...@yahoo.com wrote:

 Hi All;

 Any one advise for open source prepaid billing other than
 A2Billing that can work with Asterisk and it is rich by features
 (for large business)?

 Regards
 Bilal




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Re: [asterisk-users] Unable to insert cdr-data into mysql-DB

[Bruce Ferrell]

On 01/24/2011 07:43 AM, Jonas Kellens wrote:
 Hello list,

 I keep on getting the error :

 ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server
 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost'
 (using password: YES)


 I have a 'cdr' table in my MySQL-DB. On this table the user
 'asteriskcdr' has select, insert, update privileges.

 GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO
 'asteriskcdr'@'127.0.0.1';


 cdr_mysql.conf :

 [global]
 hostname=127.0.0.1
 dbname=Asterisk
 table=cdr
 password=mysecret
 user=asteriskcdr
 port=3306
 sock=/tmp/mysql.sock
 userfield=1

 I really don't know why Asterisk cannot connect to the table..


 Kind regards,
 Jonas.


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redo your grant like this:

GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr` TO
'asteriskcdr'@'127.0.0.1' IDENTIFIED by 'mysecret';

follow it with:

flush privileges;


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Re: [asterisk-users] UPDATE !! Spontaneous reboots on asterisk 1.6.2.11

[Bruce Ferrell]

On 09/15/2010 05:45 AM, Steve Howes wrote:
 On 15 Sep 2010, at 13:22, Jonas Kellens wrote:
   
 I have indeed found the core file in /tmp (that is where 'locate' does 
 not look huh...)
 
 'updatedb'?

 S
   
off topic, but updatedb deliberately doesn't usually look in /tmp

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Re: [asterisk-users] Registering and initiating a SIP call without a SIP client

[Bruce Ferrell]

It can be done either using a call file and a clever dial plan or via
the manager interface, again with a clever dialplan


On 09/05/2010 03:20 PM, Gautam Desai wrote:
 Can I generate SIP registration and call from Asterisk without a SIP
 client? I need to initiate a call from asterisk and play a recorded
 message.

 Gautam



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[asterisk-users] Migrating 1.4 to 1.6.2

[Bruce Ferrell]

much static testing of my realtime configuration and applications I'm
almost ready to pull the trigger.

The one thing I've been able to determine is what I need to do to
migrate my g729 licenses.

Has anyone got any advice for me on this?  The Digium site is...
difficult to navigate

TIA
Bruce Ferrell

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[asterisk-users] Looking for MIB description

[Bruce Ferrell]

Hi,

I've gone through the source tree and I don't see a MIB description file
for the SNMP agent in asterisk.  can someone point me to it.

Thanks,

Bruce ferrell

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Re: [asterisk-users] How does deny/permit work in sip.conf?

[Bruce Ferrell]

On 08/06/2010 07:45 AM, Frank Church wrote:
 I have been seeing some attempts to register devices on my Asterisk
 and I want to reconfigure it so that devices will be registered only
 if they are from the correct address, ie 192.168.1.8/255.255.255.255.

 I thought using a config like

 deny=0.0.0.0/0.0.0.0
 permit=192.168.1.8/255.255.255.255

 but it is not working the way I thought?

 Does that need a host=static.ip entry to work, rather than the
 deny/permit option?

 Does using a host=dynamic setting override any deny/permit and
 port=5060 options?

 Does being a peer or a user make a difference here?

   
I had this same problem once.  host=ip address  or host=dynamic if you
want to use permit/deny.  Permit/deny and host=dynamic allows a sip peer
or user to have a range of addresses.

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Re: [asterisk-users] How does deny/permit work in sip.conf?

[Bruce Ferrell]

On 08/06/2010 02:16 PM, Frank Church wrote:
 On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote:
   
 On 08/06/2010 07:45 AM, Frank Church wrote:
 
 I have been seeing some attempts to register devices on my Asterisk
 and I want to reconfigure it so that devices will be registered only
 if they are from the correct address, ie 192.168.1.8/255.255.255.255.

 I thought using a config like

 deny=0.0.0.0/0.0.0.0
 permit=192.168.1.8/255.255.255.255

 but it is not working the way I thought?

 Does that need a host=static.ip entry to work, rather than the
 deny/permit option?

 Does using a host=dynamic setting override any deny/permit and
 port=5060 options?

 Does being a peer or a user make a difference here?


   
 I had this same problem once.  host=ip address  or host=dynamic if you
 want to use permit/deny.  Permit/deny and host=dynamic allows a sip peer
 or user to have a range of addresses.

 --
 
 Does permit/deny  have any influence on registration, or is it related
 to the destinations it can call to or receive call from?

 How do you stop an asterisk server from accepting registrations when
 the IP is outside a subnet even if the username and secret are
 correct?

 When host=dynamic registrations are accepted even if the pemit IP is
 different from the registered device's IP address. Does permit/deny
 work on a  single IP address eg 192.168.4.111/255.255.255.2555


 The same seems to apply in the [general] section, with contactdeny and
 contacnt permit

 When I set

 contactdeny=0.0.0.0/0.0.0.0
 contactpermit=192.168.4.111/255.255.255.255

 Devices whose IP is not 192.168.4.111 are able to register.

   

When I've used permit/deny, I did it in conjunction with insecure set to
port,invite to allow gateways that didn't register and don't use
username/secret to originate calls but only from the ip range in
permit.  In fact it was for a provider that had gateways on a large
number of IP addresses, all in the same CIDR block and I didn't want to
do an entry for each of  more than 100 gateways.

contactpermit/contactdeny *should* work as you are suggesting that you
want I've never tried that.  I may attempt it tonight and see on my 1.4
system.

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Re: [asterisk-users] How does deny/permit work in sip.conf?

[Bruce Ferrell]

On 08/06/2010 07:30 PM, Bruce Ferrell wrote:
 On 08/06/2010 02:16 PM, Frank Church wrote:
   
 On 6 August 2010 16:21, Bruce Ferrell bferr...@baywinds.org wrote:
   
 
 On 08/06/2010 07:45 AM, Frank Church wrote:
 
   
 I have been seeing some attempts to register devices on my Asterisk
 and I want to reconfigure it so that devices will be registered only
 if they are from the correct address, ie 192.168.1.8/255.255.255.255.

 I thought using a config like

 deny=0.0.0.0/0.0.0.0
 permit=192.168.1.8/255.255.255.255

 but it is not working the way I thought?

 Does that need a host=static.ip entry to work, rather than the
 deny/permit option?

 Does using a host=dynamic setting override any deny/permit and
 port=5060 options?

 Does being a peer or a user make a difference here?


   
 
 I had this same problem once.  host=ip address  or host=dynamic if you
 want to use permit/deny.  Permit/deny and host=dynamic allows a sip peer
 or user to have a range of addresses.

 --
 
   
 Does permit/deny  have any influence on registration, or is it related
 to the destinations it can call to or receive call from?

 How do you stop an asterisk server from accepting registrations when
 the IP is outside a subnet even if the username and secret are
 correct?

 When host=dynamic registrations are accepted even if the pemit IP is
 different from the registered device's IP address. Does permit/deny
 work on a  single IP address eg 192.168.4.111/255.255.255.2555


 The same seems to apply in the [general] section, with contactdeny and
 contacnt permit

 When I set

 contactdeny=0.0.0.0/0.0.0.0
 contactpermit=192.168.4.111/255.255.255.255

 Devices whose IP is not 192.168.4.111 are able to register.

   
 
 When I've used permit/deny, I did it in conjunction with insecure set to
 port,invite to allow gateways that didn't register and don't use
 username/secret to originate calls but only from the ip range in
 permit.  In fact it was for a provider that had gateways on a large
 number of IP addresses, all in the same CIDR block and I didn't want to
 do an entry for each of  more than 100 gateways.

 contactpermit/contactdeny *should* work as you are suggesting that you
 want I've never tried that.  I may attempt it tonight and see on my 1.4
 system.

   

To follow up on my own reply.  I just tried this with one of my standard
peers that I use for a softphone on a 1.6.2.10  and see the registration
attempt come in at the console and a warning comes up

: Host '192.0.2.40' disallowed by contact ACL (violating IP 192.0.2.40)
: Registration denied because of contact ACL

The peer does show in sip show peers and the softphone (twinkle) shows a
Registration Fails with a 603 denied.

So I'd say it's working

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[asterisk-users] OK, I'm stumped

[Bruce Ferrell]

I'm trying to make an AMI call.  I want to call a number, play an
announcement when the call is answered, then call a second number and
connect the two when the second call is answered.

I an able to make a simple call to two numbers and connect them using
the manager API but playing the announcement has me beat.

Suggestions anyone?

Bruce Ferrell

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[asterisk-users] 1.6.2.7 SIP realtime problem

[Bruce Ferrell]

I'm getting the following message in my full log at startup and my
realtime sip peers aren't being found. My realtime extensions have no
errors.  The table sippeers exists in the database.   Is this a known
problem?

res_config_mysql.c: Table sippeers not found in database.  This table
should exist if you're using realtime.

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Re: [asterisk-users] 1.6.2.7 SIP realtime problem

[Bruce Ferrell]

On 05/14/2010 04:17 PM, Tilghman Lesher wrote:
 On Friday 14 May 2010 16:09:58 Bruce Ferrell wrote:
   
 I'm getting the following message in my full log at startup and my
 realtime sip peers aren't being found. My realtime extensions have no
 errors.  The table sippeers exists in the database.   Is this a known
 problem?

 res_config_mysql.c: Table sippeers not found in database.  This table
 should exist if you're using realtime.
 
 Check your [context] in res_mysql.conf.  In previous versions, it was set as
 [general], but extconfig.conf had asterisk as the name of the connection.
 These two configuration files need to match.  It's correct in the sample
 configs, but if you upgraded from a prior version, it's possible that you
 still have the bad match.

   
That did it.

in res_mysql it's

[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = astuser
dbpass = astpass
dbport = 3306
dbsock = /var/run/mysql/mysql.sock
requirements=warn



in extconfig the line

  sippeers = mysql,asterisk,sip_buddies

asterisk pointed to the dbname in res_mysql, not the context.

It still works that way in 1.4.31

That was fun... NOT! :)

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Re: [asterisk-users] Realtime extensions

[Bruce Ferrell]

On 02/20/2010 01:53 AM, jonas kellens wrote:
 I have read on this list that people do not get a reply if they ask
 stupid questions.

 Is this then a stupid question that I ask ?

 If nobody has ever combined extensions.conf and realtime in a way that
 I want to do, I wanna hear it too. Even if this means no solution for
 me. Then I know it's not doable.

 Jonas.

 On Thu, 2010-02-18 at 20:15 +0100, jonas kellens wrote:
 How about something like :

 [mycontext]
 exten = 100,1,NoOp(calling 100)
 exten = 100,n,NoOp(going realtime)
 switch = Realtime/mycont...@realtime_extensions
 mailto:mycont...@realtime_extensions ; from here on we use realtime

 And then my MySQL-DB contains :

 `extensions_table` VALUES (1, 'mycontext', '100', n, 'Wait', '2');
 `extensions_table` VALUES (2, 'mycontext', '100', n, 'NoOp', 'into
 RealTime');
 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback',
 'my-sound-file');

 extconfig.conf has :

 realtime_extensions = mysql,asterisk,extensions_table


 Is all the above correct and possible ??


 On Thu, 2010-02-18 at 13:55 -0500, Jared Smith wrote:
 On Thu, 2010-02-18 at 19:46 +0100, jonas kellens wrote:
  Does a context need completely be written or in extensions.conf or in
  the mysql-table 'extensions_table' ? Or can I combine the two with the
  'switch'-statement ??

 You can certainly combine the two with a switch statement.  Asterisk
 will then only look in the switch if it doesn't find a match in
 extensions.conf.

 --
 Jared Smith
 Digium, Inc.

   
You can't use the n priority construct in realtime.  the database schema
won't tolerate it and has already been mentioned even if ti would it
lacks the structure of the flat acsii file
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Re: [asterisk-users] Problems with SIP realtime

[Bruce Ferrell]

On 02/22/2010 11:13 AM, jonas kellens wrote:
 I have followed the instructions on voip-info.org for Realtime SIP
 peers, but I get this notice :

 [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889
 handle_request_register: Registration from
 'sip:test...@192.168.1.150;transport=UDP' failed for '192.168.1.105'
 - No matching peer found

 The CLI shows :

 [Feb 22 19:58:23]   == Parsing '/etc/asterisk/extconfig.conf': [Feb 22
 19:58:23] Found
 [Feb 22 19:58:23]   == Binding voicemail to
 mysql/AsteriskHosted/voicemail_users
 [Feb 22 19:58:23]   == Binding sipusers to
 mysql/AsteriskHosted/sip_buddies
 [Feb 22 19:58:23]   == Binding sippeers to
 mysql/AsteriskHosted/sip_buddies

 I have the following in extconfig.conf :

 sipusers = mysql,Asterisk,sip_buddies
 sippeers = mysql,Asterisk,sip_buddies

 I have the following in res_mysql.conf :

 [general]
 dbhost = 127.0.0.1
 dbname = Asterisk
 dbuser = asteriskuser
 dbpass = asteriskpasswd
 dbport = 3306
 dbsock = /tmp/mysql.sock

 Something I'm missing ?? Need extra configuration ?

 Jonas. 

The error message would seem to indicate that the endpoint isn't being
matched in sip_buddies.  you need the host column to be set to dynamic
for the peer.

All else seems to be in order

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[asterisk-users] two stage dialing in a SIP dial plan

[Bruce Ferrell]

I think it's called two stage dialing.

what I want is the equivalent of dialing 9995551212, waiting a fixed
time, say 1 second then dialing 4 more digits, presuming 9995551212
answers.  is this possible?  If so, how do I code it in the dialplan?

Thanks in advance

Bruce

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Re: [asterisk-users] two stage dialing in a SIP dial plan

[Bruce Ferrell]

On 01/24/2010 01:43 PM, Doug Lytle wrote:
 Bruce Ferrell wrote:
   
 I think it's called two stage dialing.

 what I want is the equivalent of dialing 9995551212, waiting a fixed
 time, say 1 second then dialing 4 more digits, presuming 9995551212
 answers.  is this possible?  If so, how do I code it in the dialplan?

 
 You need to add a wait between the numbers, I do extension look ups 
 between sites and set caller-id name, I send source destination after I 
 call their main number.:

 exten = _742XX,n,Dial(DAHDI/G2/1734xxxw${destination}${source})

 Doug

   
something like that, but I need to connect to the meetme application so
I need to dial the meetme access number then the conference.

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[asterisk-users] How to enable a range of IP addresses in realtime sip_buddies

[Bruce Ferrell]

I'm using realtime sip peers and I need to enable a range of IP
addresses for a peer.

I have:

deny  = 0.0.0.0/0.0.0.0
permit= xxx.yyy.zzz.0/255.255.255.0
mask  = 255.255.255.0
defaultIP = xxx.yyy.zzz.112
host  = xxx.yyy.zzz.112

Addresses other than .112 are being denied.  Can someone offer
assistance? Am I doing something wrong?

Bruce

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Re: [asterisk-users] How to enable a range of IP addresses in realtime sip_buddies

[Bruce Ferrell]

Thank you!

Jonathan Thurman wrote:
 You need to set: host=dynamic  Otherwise only .112 is allowed.
 
 -Jonathan
 
 
 On Tue, Jan 19, 2010 at 1:17 PM, Bruce Ferrell bferr...@baywinds.org wrote:
 I'm using realtime sip peers and I need to enable a range of IP
 addresses for a peer.

 I have:

 deny  = 0.0.0.0/0.0.0.0
 permit= xxx.yyy.zzz.0/255.255.255.0
 mask  = 255.255.255.0
 defaultIP = xxx.yyy.zzz.112
 host  = xxx.yyy.zzz.112

 Addresses other than .112 are being denied.  Can someone offer
 assistance? Am I doing something wrong?

 Bruce

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[asterisk-users] DAHDI - Split data voice use

[Bruce Ferrell]

Can any Digium E1 cards be used for split data/voice use?

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Re: [asterisk-users] Possibly I don't understand sip peers

[Bruce Ferrell]

Andrew Thomas wrote:
 [peer]
 defaultip=xxx.xxx.xxx.xxx
 host=xxx.xxx.xxx.xxx
 deny=0.0.0.0/0.0.0.0
 
 allow=xxx.xxx.xxx.0/255.255.255.0  read what you've put!!!  The
 'allow' should be 'permit' as Jared already told you (and he should know
 what he's talking about).
 
 insecure=port,invite

 
 
 
   
   
 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce
 Ferrell
 Sent: 29 July 2009 23:34
 To: jsm...@digium.com; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [asterisk-users] Possibly I don't understand sip peers
 
 
 
 Jared Smith wrote:
 On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
 I have a carrier who tells me he will be sending me traffic from a
 wide
 range of IP addresses.

 so I set up a realtime peer as follows:

 [peer]
 defaultip=xxx.xxx.xxx.xxx
 host=xxx.xxx.xxx.xxx
 deny=0.0.0.0/0.0.0.0
 allow=xxx.xxx.xxx.0/255.255.255.0
 insecure=port,invite


 Yes, he's really claiming to originate from any of the IP in the
 block
 When I leave the host blank, we reject calls with a 404.

 shouldn't I be able to put in a kind of wildcard for his IP block
 or
 am I just being silly?  If not, what am I doing wrong?
 I think you've got your syntax wrong there... permit and deny
 statements are used to create Access Control Lists and to limit the IP
 address ranges.  The allow and disallow statements are to allow or
 disallow various codecs.  They way you've specified it above, you're
 allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably
 isn't what you want.


 
 I have the codec permissions in the columns allow and disallow.  Those
 seem to work ok.
 
 it's permit/deny/mask I seem to be having a problem with.  Like I say, I
 don't think I understand their use or perhaps they don't work in
 realtime
 

Now that we're done making fun of me because I mis entered allow/permit,
can we address what I was trying to ask...

deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.xxx/255.255.255.0
mask=255.255.255.0

if host is set to a specific IP in  the range specified by permit, calls
are rejected if not from that specific IP.
if host is set to dymanic, calls are rejected.
defaultip is set to the specific IP

what do I not understand?

I really appreciate your help, even the snarky stuff

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Re: [asterisk-users] Possibly I don't understand sip peers

[Bruce Ferrell]

Jared Smith wrote:
 On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
 I have a carrier who tells me he will be sending me traffic from a wide
 range of IP addresses.

 so I set up a realtime peer as follows:

 [peer]
 defaultip=xxx.xxx.xxx.xxx
 host=xxx.xxx.xxx.xxx
 deny=0.0.0.0/0.0.0.0
 allow=xxx.xxx.xxx.0/255.255.255.0
 insecure=port,invite


 Yes, he's really claiming to originate from any of the IP in the block

 When I leave the host blank, we reject calls with a 404.

 shouldn't I be able to put in a kind of wildcard for his IP block or
 am I just being silly?  If not, what am I doing wrong?
 
 I think you've got your syntax wrong there... permit and deny
 statements are used to create Access Control Lists and to limit the IP
 address ranges.  The allow and disallow statements are to allow or
 disallow various codecs.  They way you've specified it above, you're
 allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably
 isn't what you want.
 
 

I have the codec permissions in the columns allow and disallow.  Those
seem to work ok.

it's permit/deny/mask I seem to be having a problem with.  Like I say, I
don't think I understand their use or perhaps they don't work in realtime



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Re: [asterisk-users] Possibly I don't understand sip peers

[Bruce Ferrell]

Anthony wrote:
 Jared Smith wrote:
 On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
   
 I have a carrier who tells me he will be sending me traffic from a wide
 range of IP addresses.

 so I set up a realtime peer as follows:

 [peer]
 defaultip=xxx.xxx.xxx.xxx
 host=xxx.xxx.xxx.xxx
 deny=0.0.0.0/0.0.0.0
 allow=xxx.xxx.xxx.0/255.255.255.0
 insecure=port,invite


 Yes, he's really claiming to originate from any of the IP in the block

 When I leave the host blank, we reject calls with a 404.

 shouldn't I be able to put in a kind of wildcard for his IP block or
 am I just being silly?  If not, what am I doing wrong?
 
 I think you've got your syntax wrong there... permit and deny
 statements are used to create Access Control Lists and to limit the IP
 address ranges.  The allow and disallow statements are to allow or
 disallow various codecs.  They way you've specified it above, you're
 allowing a codec called xxx.xxx.xxx.0/255.255.255.0, which probably
 isn't what you want.


   
 
 Your looking for host=dynamic.
 
 
 Anthony
 

Tried that.  dynamic seems to require a registration to work.  Carriers
don't register.



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[asterisk-users] Possibly I don't understand sip peers

[Bruce Ferrell]

I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.

so I set up a realtime peer as follows:

[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
allow=xxx.xxx.xxx.0/255.255.255.0
insecure=port,invite


Yes, he's really claiming to originate from any of the IP in the block

When I leave the host blank, we reject calls with a 404.

shouldn't I be able to put in a kind of wildcard for his IP block or
am I just being silly?  If not, what am I doing wrong?

Bruce

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Re: [asterisk-users] Fwd: Unknown udp ports listening experts calling !

[Bruce Ferrell]

Xavier Cardil wrote:
 I found nothing is passing through those ports . . . I think something
 was sending the stream to our PST/SIP gateways, so the calls where
 affected when getting in to the gateways. I found we are not running any
 extra TCL applications on those gateways . . . could it be possible ?
 Could an UDP stream get mixed with another through an UDP port ? Is a
 very strange issue but I really want to know why . . . any more hints ?
 
 Thanks.
 
 On Wed, Jul 1, 2009 at 11:48 AM, John A. Sullivan III
 jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com
 wrote:
 
 On Wed, 2009-07-01 at 10:14 +0100, Steve Howes wrote:
  On 1 Jul 2009, at 09:54, Xavier Cardil wrote:
   udp0  0 0.0.0.0:2727 http://0.0.0.0:2727
   0.0.0.0:*   4989/asterisk
   udp0  0 0.0.0.0:9001 http://0.0.0.0:9001
   0.0.0.0:*   26354/udp-sender
   udp0  0 0.0.0.0:5000 http://0.0.0.0:5000
   0.0.0.0:*   4989/asterisk
 
  2727 = mgcp
 
  I found that with Google. A useful tool.
 snip
 I thought 9001 was for JetDirect style print servers.  I don't recall
 off the top of my head if they are tcp or udp - John
 --
 John A. Sullivan III
 Open Source Development Corporation
 +1 207-985-7880
 jsulli...@opensourcedevel.com mailto:jsulli...@opensourcedevel.com
 
 http://www.spiritualoutreach.com
 Making Christianity intelligible to secular society
 


Assuming first your box doesn't have a rootkit installed  (to check for
a rootkit, use rkhunter.  Your distro may have it packaged, if not
google for it) I use lsof to find out what is listening to TCP and UDP
ports:

lsof -P | grep UDP
lsof -P | grep TCP

YMMV

Bruce



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Re: [asterisk-users] Asterisk, SQL Database Update

[Bruce Ferrell]

Edwin Quijada wrote:
 
 Thanks for your helpful reply.



 I am not so good in coding.



 simply all i need is as follow



 When a call comes, goes into an IVR, and then depending on the entry option

 it will connect to a remote SQL Database, to check the pre-existed data,

 and in the end of the IVR the caller will enter an option that will need to 
 be written in the SQL Database.



 Can you please give me a general scenrio how this will be achieved.

 and which files that i will need to modify.

 
 I think that if you are not good coding you will have a few problems.
 Maybe, the best solution 4u is hire external to do that. It is simple 
 but just in dialplan it is so difficult with AGI it is so easy but
 you dont want coding.
  

depending on what you have in mind the MYSQL command is quite useful.
Have a look at the examples here for some ideas:

http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL

I'm known far and wide as a gonzo hack, but it usually get's the task
accomplished.  :)




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[asterisk-users] Wanting to manipulate SIP response headers

[Bruce Ferrell]

My boss has asked me if there is a way to send back a 503 response to a
request at will.

I don't see anything in the documents that would allow for manipulation
in asterisk at that low of a level.

Am I wrong?

Bruce Ferrell

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Re: [asterisk-users] Wanting to manipulate SIP response headers

[Bruce Ferrell]

Philipp Kempgen wrote:
 Bruce Ferrell schrieb:
 My boss has asked me if there is a way to send back a 503 response to a
 request at will.

 I don't see anything in the documents that would allow for manipulation
 in asterisk at that low of a level.

 Am I wrong?
 
 Depends.
 Hangup(34) would send a 503 Service Unavailable on the SIP channel.
 See hangup_cause2sip() in chan_sip.c
 
 But in a way you are right. Hangup() works on the call level so
 you can't simply tell Asterisk to send a 503 response to an arbitrary
 request.
 
 
 Philipp Kempgen

That will actually serve to do what I need.  Thanks!

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Re: [asterisk-users] Asterisk cmd MYSQL app_addon_sql_mysql / performance ?

[Bruce Ferrell]

There is no perl instance used by MYSQL.  It uses the mysql client library

Julien Chavanton wrote:
 I was looking for a (http socket module / mysql module) not using
 AGI(perl/php/shell) for asterisk in order to do intensive database / web
 server interactions as needed without performance to much overhead.
  
 Is there a real benefit in using : Asterisk cmd MYSQL
 app_addon_sql_mysql , I see it require perl DBD so it is not using Mysql
 C API, then there must be an instance of the perl interpreter for every
 query ?
  
  
  
  
 
 
 
 
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Re: [asterisk-users] H323plus homepage down?

[Bruce Ferrell]

Sebastian wrote:
 Anybody knows why is down? Or if has been moved to another page??
 
  
 
 
 
 
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It's up for me

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[asterisk-users] Can't compile Dahdi on SuSE 10.1/Kernel 2.6.16.54-0.2.5-smp/GCC 4.1.2

[Bruce Ferrell]

Below is the output when the compile fails.  any suggestions?



make -C /lib/modules/2.6.16.54-0.2.5-smp/build ARCH=i386 
SUBDIRS=/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi 
DAHDI_INCLUDE=/usr/src/dahdi-linux-2.1.0.4/include DAHDI_MODULES_EXTRA= 
 HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
make[1]: Entering directory `/usr/src/linux-2.6.16.54-0.2.5-obj/i386/smp'
make -C ../../../linux-2.6.16.54-0.2.5 
O=../linux-2.6.16.54-0.2.5-obj/i386/smp modules
   CC [M]  /usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_dynamic_eth.o
/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_dynamic_eth.c: In 
function 'ztdeth_rcv':
/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_dynamic_eth.c:104: 
error: too many arguments to function 'skb_linearize'
make[4]: *** 
[/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi/dahdi_dynamic_eth.o] Error 1
make[3]: *** [_module_/usr/src/dahdi-linux-2.1.0.4/drivers/dahdi] Error 2
make[2]: *** [modules] Error 2
make[1]: *** [modules] Error 2
make[1]: Leaving directory `/usr/src/linux-2.6.16.54-0.2.5-obj/i386/smp'
make: *** [modules] Error 2




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[asterisk-users] Dial weirdness

[Bruce Ferrell]

I'm seeing this response to SIP calls originated in the following manner:

Dial(SIP/${EXTEN}SIP/{$DID},30,r)

handle_response_invite: Re-invite to non-existing call leg on other UA.

The response is from the second part of the dial.  What exactly does it 
mean and how can I fix it?

Thanks in advance

Bruce

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Re: [asterisk-users] Allison Smith, Music-on-Hold Parody--outstanding.

[Bruce Ferrell]

Karl Fife wrote:
 Allison Smith just created a hysterical parody music on hold Parody.  
 Whatever you were doing, stop, and dial this number to listen to it: 
 360-519-5689. 2 minutes.
  
 I just gave her a few ideas, but she took it and ran with it--she chose 
 the audio and did the mix-down and everything.  Really funny!! 
  
 -Karl 
  

It's like something from Mad Magazine

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Re: [asterisk-users] how to get /var/run/asteris/asterisk.ctl

[Bruce Ferrell]

Scott Berry wrote:
 Hello there everyone,
 
 Well I have set up Asteriks 6.0 and almost have Freepbx working too.
 However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is
 not found.  I confirmed that by going to the directory.  How do I
 get /var/run/asterisk/asterisk.ctl put in correctly?  I am using a
 Ubuntu 8.10 system.  Thanks much.
 

When you start asterisk, it creates the control socket.  Have a look at 
/etc/asterisk/asterisk.conf to see if your install has put it somplace 
other than the stated default location.  If you think asterisk is 
started and the control socket isn't there, look at 
/var/log/asterisk/full to see why it didn't start

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[asterisk-users] Realtime and OOH323

[Bruce Ferrell]

Can Realtime be used with OOH323 ala sip_buddies?

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Re: [asterisk-users] CDR

[Bruce Ferrell]

Andrew Kohlsmith wrote:
 On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote:
 Michael Collins wrote:
 I don't know if it's relevant or not, but I do know that at least one
 legacy PBX vendor (NEC) has a 'solution' that helps with some of the
 sillier CDR's that could get generated.  They have what they call a
 pseudo-answer timer which is basically just a way of saying, If a
 call doesn't last for at least X number of seconds then it really isn't
 a call and no CDR should be generated.  It is a bit of a case of
 throwing away all really short phone calls, even legit ones, but it does
 also get rid of the silly stuff: I pick up, get dial tone, then hang up
 or I pick up, dial ext 1234, let it ring for two seconds and then hang
 up.
 I would definitely want a CDR record in this case.
 
 Out of curiosity... why? Both you and Atis seem to want to see CDRs for 
 non-calls, and I'm unable to see why.
 
 -A.

Well... I'm just guessing, but lacking any other means of collecting
information about call attempts, CDRs got elected.

When I worked on a DMS250 we had operational measurements that would
tell us about such things and CDRs were only for billing.



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Re: [asterisk-users] CallerID problem Asterisk 1.4.2

[Bruce Ferrell]

try ${CALLERID(all)}

Peter Kranz wrote:
 When receiving inbound calls from a Vonage Softphone extension, I'm unable
 to view/maniupulate calledid data. but it shows up in the CDR records and on
 called handsets.. any ideas?
 
 exten = asda,n,NoOp(callerID is ${CALLERID})
 exten = asda,n,NoOp(CallerID is ${CALLERIDNAME})
 exten = asda,n,NoOp(CallerID is ${CALLERIDNUM})
 
 -- Executing [EMAIL PROTECTED]:2] Wait(SIP/asd1-086775b8, 1) in new 
 stack
 -- Executing [EMAIL PROTECTED]:3] NoOp(SIP/asd-086775b8, callerID is 
 ) in
 new stack
 -- Executing [EMAIL PROTECTED]:4] NoOp(SIP/asd-086775b8, CallerID is 
 ) in
 new stack
 -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/asd-086775b8, CallerID is 
 ) in
 new stack
 
 But CDR data is there.
 
 WIRELESS CALLER
 asd,asd,100,pstn-in,SIP/asd-086775b8,SIP/peter5-08689458,Dial
 ,SIP/peterS
 IP/peter2SIP/peter3SIP/peter4SIP/peter5|25|m,2007-09-24
 11:19:27,2007-09-24 11:19:27,2007-09-24 11:19:44,17,17,ANSWER
 ED,DOCUMENTATION,,1190657967.4165,
 
 Peter Kranz
 Founder/CEO - Unwired Ltd
 www.UnwiredLtd.com
 Desk: 510-868-1614 x100
 Mobile: 510-207-
 [EMAIL PROTECTED]
 
 
 
 
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Re: [asterisk-users] where is 1.4.12?

[Bruce Ferrell]

Stephen Bosch wrote:
 Jared Smith wrote:
 On Thu, 2007-08-30 at 08:02 -0500, Eric ManxPower Wieling wrote:
 As I understand it, Digium does NO formal QA testing before the free 
 Asterisk/Zaptel/libPRI releases.  Asterisk Business Edition is a 
 different story and gets extensive QA testing.
 As I understand it, that's simply due to a lack of resources.  At the
 Asterisk Developer's Conference earlier this year, the Asterisk
 Developers were all pretty much in agreement that more needed to be done
 in this area, but that it would have to be a combined effort between the
 Asterisk community and Digium, as Digium simply doesn't have the
 resources at this point to do it all itself.

 On IRC I have been a vocal user from hell about the QA issues of 
 Digium open source products.  
 I've tried to be vocal about this too.  And now that I'm working for
 Digium, I'd be happy to try to coordinate an effort between the
 community and Digium to try to come up with a framework where we can all
 work together to make this happen.  
 
 That's the spirit.
 
 I just wanted to throw in something.
 
 That a product is commercial is never an assurance that it is or will be
 stable. It doesn't matter if the product is from IBM (which spends a
 billion dollars on RD annually) or Cisco or Nortel.
 
 Commercial products break, no matter how much they cost.
 
 Most vendors do a careful job of obfuscating the instability in their
 own product. Depending on the depth in their technical staff, they will
 solve the problem quickly or slowly, or offer you some limp workaround.
 That is somewhat correlated with the cost and class of the product. If
 you believe, however, that paying for a product means that it will work
 reliably or as promised, you are living in a Madison Avenue-induced haze.
 
 Either way... it shouldn't be an excuse for us or Digium to accept less.
 Take the Linux kernel -- there is a community project with a rigorous
 vetting process, and I would say the Linux kernel is extremely stable.
 
 We can and should introduce a similar rigor for Asterisk. A big step in
 that direction is patience and focus. The creeping featurism could
 make way for an increased concentration on reliability. That's a
 development roadmap thing.
 
 Cheers,
 
 -Stephen-
 

I started this whole thing by misreading an announcement.

I can't get the core information right not as I rolled back to keep my
office happy.  Over the weekend I'll reload 1.4.11 and wait for a core
dump.  they happen regularly enough :(  then I'll get a stack trace and
whatever else is needed into the bug tracker to solve the problem.

Given that there are a couple of days before I can do this, anyone care
to give me some pointers as to what I should collect so I can get it all
in one go?

Thanks





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[asterisk-users] where is 1.4.12?

[Bruce Ferrell]

Several days ago an announcement came out for a SIP bug in versions
below 1.4.12.  So far I don't see 1.4.12 available for download and I'm
seeing something that may be the bug...  My asterisk is restarting
itself about every 30 minutes.

HELP!!!

:)

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Re: [asterisk-users] extension.conf doesn't reload?

[Bruce Ferrell]

Devraj Mukherjee wrote:
 Hi everyone,
 
 I have just installed Asterisk 1.4.6 on CentOS 5. When I issues the
 reload command in the asterisk command prompt, it doesn't seem to read
 my configuration files. Any suggestions?
 
 pbx*CLI reload
 The 'reload' command is deprecated and will be removed in a future
 release. Please use 'module reload' instead.
   == Parsing '/etc/asterisk/cdr.conf': Found
 [Jul 23 14:14:54] NOTICE[28392]: cdr.c:1359 do_reload: CDR simple
 logging enabled.
   == Parsing '/etc/asterisk/dnsmgr.conf': Found
   == Parsing '/etc/asterisk/manager.conf': Found
   == Parsing '/etc/asterisk/rtp.conf': Found
   == RTP Allocating from port range 1 - 2
   == Parsing '/etc/asterisk/http.conf': Found
 
 
Try dialplan reload

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Re: [asterisk-users] Gtalk/Jabber connect issues in 1.4.8

[Bruce Ferrell]

DOH!  Thanks

[EMAIL PROTECTED] wrote:
 Hi,
 
 That should be:
 
 [EMAIL PROTECTED]
 
 Got it?
 
 
 BR,
 Dems
 
 
I've included my jabber.conf below.  I'm betting the following errors:

[Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER:
Node Error
[Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop:
JABBER: Got hook event.

jabber test

[Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send:
JABBER: Not connected can't send
User: [EMAIL PROTECTED]

Anyone have a hint??

Thanks

[general]
debug=no;;Turn on debugging by default.
autoprune=yes   ;;Auto remove users from buddy list.
autoregister=yes;;Auto register users from buddy
list.

[asterisk]  ;;label
type=client ;;Client or Component connection
 ;;Route to server for example,
serverhost=talk.google.com
 ;;  talk.google.com
;[EMAIL PROTECTED]/asterisk  ;;Username with optional roster.
[EMAIL PROTECTED]
secret= ;;Password
port=5222   ;;Port to use defaults to 5222
usetls=yes  ;;Use tls or not
usesasl=yes ;;Use sasl or not
[EMAIL PROTECTED];;Manual addition of buddy
statusmessage=I am available  ;;Have custom status message for
 ;;Asterisk.
timeout=100 ;;Timeout on the message stack.

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[asterisk-users] Gtalk/Jabber connect issues in 1.4.8

[Bruce Ferrell]

I've included my jabber.conf below.  I'm betting the following errors:

[Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER: 
Node Error
[Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop: 
JABBER: Got hook event.

jabber test

[Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send: 
JABBER: Not connected can't send
User: [EMAIL PROTECTED]

Anyone have a hint??

Thanks

[general]
debug=no;;Turn on debugging by default.
autoprune=yes   ;;Auto remove users from buddy list.
autoregister=yes;;Auto register users from buddy 
list.

[asterisk]  ;;label
type=client ;;Client or Component connection
 ;;Route to server for example,
serverhost=talk.google.com
 ;;  talk.google.com
;[EMAIL PROTECTED]/asterisk  ;;Username with optional roster.
[EMAIL PROTECTED]
secret= ;;Password
port=5222   ;;Port to use defaults to 5222
usetls=yes  ;;Use tls or not
usesasl=yes ;;Use sasl or not
[EMAIL PROTECTED];;Manual addition of buddy 
statusmessage=I am available  ;;Have custom status message for
 ;;Asterisk.
timeout=100 ;;Timeout on the message stack.

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[asterisk-users] Asterisk 1.4.4 Segfaults with asterisk-ooh323 from addons-1.4.1

[Bruce Ferrell]

the config file is basically the sample file

gdb of the core files show the below.

Loaded symbols for /usr/lib/asterisk/modules/chan_ooh323.so
#0  0xb7d203e7 in strcasecmp () from /lib/libc.so.6


Any suggestions?
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Re: [asterisk-users] linksys pap2 version2 ata DTMF issue

[Bruce Ferrell]

Troy Ayers wrote:
My asterisk box doesn't recognize DTMF from my analog phone, plugged 
into my ATA(linksys pap2 version2).


I can make/receive calls fine... it's just that, for example, I cannot 
login to my asterisk voicemail.


Softphones (such as x-lite) are fine.
I've turned up a few articles via google where some people have this 
trouble, but have not seen suggestions on how to fix.  I presume this is 
an ATA problem, and I expect not much is tweakable on the Asterisk side 
regarding this, but still I am looking for suggestions for either 
Asterisk or the ATA.


-Troy



Can you access IVR systems outside of asterisk?  I've found that when I 
can then voicemail doesn't work.  When voicemail works, outside IVR doesn't.

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[asterisk-users] why do I get this message

[Bruce Ferrell]

set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only

I have 1.2.17 on Suse 10.1
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Re: [asterisk-users] why do I get this message

[Bruce Ferrell]

Alex Balashov wrote:

On Fri, 20 Apr 2007, Bruce Ferrell said something to this effect:



set_format: Unable to find a codec translation path from ulaw to g729

Both endpoints are PAP2 set to G711 only



  Where precisely are they so set?

--
Alex Balashov [EMAIL PROTECTED]


The are locked to G711u at the line... Using the web interface
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Re: [asterisk-users] Erased log files

[Bruce Ferrell]

logger reload at the CLI

Luis Claudio Santos wrote:

People,
 
I've erased the messages and full files in /var/log/asterisk/. I've 
already created other files and changed the owner, etc, and permissions:


-rw-r--r--  1 asterisk asterisk   0 Mar  2 16:01 event_log
-rw-r--r--  1 asterisk asterisk 1514385 Mar 27 18:15 full
-rw-r--r--  1 asterisk asterisk  396170 Mar 27 18:20 messages
-rw-r--r--  1 asterisk asterisk1102 Mar  3 18:08 queue_log
But, the asterisk is not writing in these files anymore.
 
Is there a solution?
 
Thanks.


--
Abraços
Luis Claudio
 





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Re: [asterisk-users] cause 127

[Bruce Ferrell]

a quick google search for isdn cause codes gives this:

http://72.14.253.104/search?q=cache:mTG1cVFR5hIJ:www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf+isdn+cause+codeshl=enct=clnkcd=3gl=us

Used to report a protocol error event only when no other cause in the 
protocol error class applies.


127
7F

Interworking, Unspecified.
Indicates that there has been interworking with a network which does not 
provide causes for actions it
takes. Thus, the precise cause for a message which is being sent cannot 
be ascertained.



Thomas Stein wrote:

Hello.

Someone knows what cause 127 mean. The phone that i'm calling rings once and 
than the connection interrupts:


P[ 5]  -- l3id:10040
P[ 5]  -- cause:127
P[ 5]  -- out_cause:127
P[ 5]  -- state:ALERTING
P[ 5]  -- Channel: mISDN/5-1 hanguped new state:CLEANING
P[ 5] $$$ CLEANUP CALLED pid:3

best regards

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[asterisk-users] OS X Frequent console disconnects 1.4.1

[Bruce Ferrell]

Hi,

I'm seeing the following message in the full log:

 WARNING[478] asterisk.c: poll returned  0: Bad file descriptor

it's repeated a number of times then I'm disconnected from the running 
asterisk instance.


What's the best way to correctly report this?
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Re: [asterisk-users] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing

[Bruce Ferrell]

Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any 
royalties to use G.729b in Romania, so I should have used other drivers. 
The installation procedure looked difficult so I decided to get one from 
Digium - it's not that expensive, my time is much more expensive.


Made the payment, got they key, downloaded and copied everything as in 
http://kb.digium.com/entry/30/5/; but when I called register I got no 
result. Actually I do get the prompt asking me to use -l to see the 
licence, nothing after that. It gives no error message, nothing at all!


My first ethernet device is eth0 so it's not that; I'm able to browse 
https sites so the ports are open. I *disabled* the firewall and tried 
again, no success, so it's not firewall related. What to do next?


Thanks.
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Contact [EMAIL PROTECTED]

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[asterisk-users] Realtime voicemail passwords

[Bruce Ferrell]

I've been looking through everything I can find and observing the mysql 
logs and I don't see password changes passing through to the DB.  Is 
that correct?


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[asterisk-users] #include not working in 1.4

[Bruce Ferrell]

I use a lot of included files in my dialplan and I see that #include 
file.conf is now interpreted as an attempt to include a non existant 
context.


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Re: [asterisk-users] Parsing Area Code from CallerID

[Bruce Ferrell]

John French wrote:
How would I parse the area code from this variable? Number=2515551212  
Sorry for the dense question, I don't seem to be able to find an 
appropriate function for parsing left to right.

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NPA=${NUMBER:0:3}

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Re: [asterisk-users] CallerID Issue (asterisk newbie)

[Bruce Ferrell]

Anselm Martin Hoffmeister wrote:

Am Mittwoch, den 13.12.2006, 15:03 + schrieb
[EMAIL PROTECTED]:

Hi guys. This is my 1st post here (after much reading). I have a test  
asterisk system setup using X-Lite Soft Phones, and the issue I am  
running into is that caller id shows up as asterisk on all incoming  
calls and on all local to local calls (internal). I have showcallerid,  
etc. configured in zapata.conf, but I'm drawing a blank.  When I check  
my voicemails it tells me that the message is from an unknown caller.   
I would appreciate any info.



Zapata.conf is not usually related to sip device callerid, if you have
no Zap interface.

Try setting the callerid= stuff in sip.conf appropriately.
Mine looks like this, for my desktop phone:

[sip504]
mailbox=01
callerid=504
type=friend
username=sip504
secret=YouDontKnowThis
context=sipclient
host=dynamic
nat=yes
disallow=all
allow=alaw
allow=gsm

so the callerid= line is what you should adapt.

BR
Anselm


I've seen this recently when the caller ID comes in 
NPANXXy somehow the very long callerid isn't handled 
well.   I ended up peeling off the first 10 digits and re-stuffing the 
callerid with that

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[asterisk-users] Interesting CALLERID behavior

[Bruce Ferrell]

I've found an interesting behavior in callerid handling.

I have very long callerid coming in or maybe just improperly combined 
information.  In any case the result is that the caller ID is set to 
asterisk on the outgoing leg.  Has anyone else seen this before?  Is 
there a solution for it?



Thanks in advance

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[asterisk-users] DTMF problems with IVR - What DMTF Tx method

[Bruce Ferrell]

I'm having problems with a new asterisk PBX install.  the phones/ATAs 
are all linksys/cisco.  They all worked before with a commercial softswitch.


Most of the linksys devices offer auto, inband, INFO and AVT.   I'm 
looking for suggestions.


Thanks in advance

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Re: [asterisk-users] echo with spa-3000

[Bruce Ferrell]

check the impedance for the fxo port.  I bet it's set to 600.  change it 
to 900 and see what happens


James Harper wrote:

More an echo algorithm question than a purely asterisk one...

I have the following setup:

Handset - PAP2 - Asterisk - SPA3000 - Telco

And no matter what I do, I get echo on a call routed out via the PSTN
when I talk into the handset, in the order of a hundred ms (my estimate,
could be wildly inaccurate!). Echo will occur also when I have a handset
plugged into the phone port on the SPA3000 (only when the call is routed
via SIP of course), but I think it's more noticeable on the PAP2 due to
the increased latency.

The echo occurs even when I call an automated voice service or
something, so I'm thinking that possibly there is a gross impedance
mismatch on my side of the telco switch. I believe the following is
happening (latency measures are guesses):

Handset -(0ms)- PAP2 -(40ms)- Asterisk -(40ms)- SPA3000 -(0ms)- Telco

Does an EC algorithm need a measurable delay to work? The EC would have
to cope with an almost unmeasurably small echo delay (the delay only
creeps in on the other side of the IP link. Is there another way I
should be solving this problem, especially as I can keep Asterisk in the
loop if I want to?

Thanks

James

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Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060

[Bruce Ferrell]

Just restart asterisk

Crazy Boy wrote:

Hi,

I am Chandra. I have a doubt related to ports. I have seen my port 5060 
status with nmap command and it is showing that 5060 blocked. Afterthat, 
I stopped firewall also. After stopping the firewall also, it is showing 
the 5060 port is blocked. Can I need to restart the linux system from 
boot to take the effect? Please tell me how to open 5060 port? Looking 
forward to your response. Thank you.


Regards,
Chandra.

Steven Ringwald [EMAIL PROTECTED] wrote:

Elpidio Ramos wrote:
  Bob,
 
  I get the same answer you get when using netstat -an
 
  When I query the firewall rules I get this:
  Chain RH-Firewall-1-INPUT (2 references)
  target prot opt source destination
  ACCEPT all -- anywhere anywhere
  ACCEPT icmp -- anywhere anywhere icmp any
  ACCEPT ipv6-crypt-- anywhere anywhere
  ACCEPT ipv6-auth-- anywhere anywhere
  ACCEPT udp -- anywhere 224.0.0.251 udp dpt:5353
  ACCEPT udp -- anywhere anywhere udp dpt:ipp
  ACCEPT all -- anywhere anywhere state
  RELATED,ESTABLISHED
  ACCEPT tcp -- anywhere anywhere state NEW
  tcp dpt:ssh
  ACCEPT tcp -- anywhere anywhere state NEW
  tcp dpt:http
  REJECT all -- anywhere anywhere
  reject-with icmp-host-prohibited
 
  I assume this indicates port 5060 is restricted?


Yep.


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Re: [asterisk-users] ASTERISK NOT LISTENING IN PORT 5060

[Bruce Ferrell]

or get really radical:

iptables -P INPUT ACCEPT
iptables -P OUTPUT ACCEPT
iptables -F


Bob Chiodini wrote:
I think all anywhere should allow 5060.  Try running service iptables 
stop (as root) to shutdown the firewall.  See if 5060 then answers.


I'm not running a firewall on my asterisk box so I'm not sure what the 
rule would need to be.  service iptables start will restore the firewall.


Bob...

Elpidio Ramos wrote:


Bob,
 
I get the same answer you get when using netstat -an
 
When I query the firewall rules I get this:

Chain RH-Firewall-1-INPUT (2 references)
target prot opt source   destinationACCEPT 
all  --  anywhere anywhere   ACCEPT icmp --  
anywhere anywhereicmp any
ACCEPT ipv6-crypt--  anywhere anywhere   
ACCEPT ipv6-auth--  anywhere anywhere   
ACCEPT udp  --  anywhere 224.0.0.251 udp dpt:5353

ACCEPT udp  --  anywhere anywhereudp dpt:ipp
ACCEPT all  --  anywhere anywherestate 
RELATED,ESTABLISHED
ACCEPT tcp  --  anywhere anywherestate NEW 
tcp dpt:ssh
ACCEPT tcp  --  anywhere anywherestate NEW 
tcp dpt:http
REJECT all  --  anywhere anywhere
reject-with icmp-host-prohibited
 
I assume this indicates port 5060 is restricted?
 
Elpidio


*/Bob Chiodini [EMAIL PROTECTED]/* wrote:

Elpidio,

Is it truly not listening or is maybe a firewall blocking port 5060.
What does netstat -an | grep 5060 tell you? I get this:

netstat -an | grep 5060
udp 0 0 0.0.0.0:5060 0.0.0.0:*

iptables -L will list any firewall restrictions.

Bob...


On Fri, 2006-09-01 at 09:12 -0700, Elpidio Ramos wrote:
 Does anyone knows what could be the cause for asterisk not 
listening

 in post 5060 if SIP interfaces is loaded with no problems?

 I am using Fedora Core 3.

 I have followed the instructions in several tutorials and tried
 several soft phones and the SIP interface seem to be dead.

 1. When loading asterisk SIP load with no problem
 2. When I activate the DEBUG for a peer, ip or sip in general, I
don't
 get to see any messages when a connection is attempted from any 
soft

 phone.
 3. The soft phones all report a timeout when trying to register.
 4. Tried also to move to port 80 that I know is open but still the
 same problem.

 I will appreciate any help anyone can provide with this problem.

 Elpidio
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*Elpidio Ramos*
President RM International Services SA CV
Web: http://www.ramosoft.com
Mex:  +52 (55) 5116-9804 Office
 +52 (55) 5116-9805 Fax
 +52 (55) 1755-6601 Cell
USA: +1 (801) 494-1415 Office
 +1 (240) 250-8264 Fax
 +1 (801) 938-4740 Direct
 



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Re: [asterisk-users] Asterisk with PABX

[Bruce Ferrell]

Just post it.  be sure to wear asbestos.  someone is sure to take 
offense.  someone else just as surely will silently find it useful


Ira wrote:

At 10:20 AM 8/30/2006, you wrote:
Well, I started writing a tutorial for programming dial plans and sent 
it to two people who claimed interest, never heard back from either so I 
stopped.  It's hard to know if what I write would be useful to anyone, 
so I don't want to just post it without feedback. I'd be happy to keep 
going if it seemed there was a point. I see the world quite different 
from the average person and it's not always clear how I explain thing is 
useful to others. I'd hate to post something that just ended up 
confusing people.


Ira


How about creating some documentation?

be to just start using *.   And again, I'm not trying to knock
TrixBox, but it's a box and if you fit in it, it's probably the best 
thing
there is. Personally, when I started, it seemed too small and limiting 
so I

learned to program my own dial plans.  It's not trivial as the language
we've been given is not really well thought out and poorly documented at
best, but in the end it's done everything I've asked of it.

Ira



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[asterisk-users] How many digits are collected

[Bruce Ferrell]

When Background or Playback is used in a dial plan how many digits are 
collected and what variable are they returned in?


I'm trying to do a simple auto attendant and having very little luck

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[asterisk-users] an operational scenario

[Bruce Ferrell]

I'm trying to do something I've not see written up here before.  I have 
an asterisk on a box with 2 interfaces like the drawing below.  I want 
to have SIP extensions regsitering to both interfaces and able to 
communicate.  Is this possible?  What suggestions do you have?



  +-+
  | |
internal  | | external
--+ +-
192.168.1 | | real IP
  | |
  +-+
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Re: [asterisk-users] an operational scenario

[Bruce Ferrell]

the problem I'm seeing is one way audio between extensions.  I've splpit 
up the numbering plan internal/external.  All are in the same range. 
I'll try splitting them and see what happens.


Erick Perez wrote:

Why can't you do it?
I have an internal 192.168.100.x address (eth0) and 200.x.x.x (eth1)
interface. Internal users register to the 192 and internet users
register to the 200.x address

internal extensions are 1XXX and external extensions are 2XXX

What errors do you have?

On 7/12/06, Bruce Ferrell [EMAIL PROTECTED] wrote:


I'm trying to do something I've not see written up here before.  I have
an asterisk on a box with 2 interfaces like the drawing below.  I want
to have SIP extensions regsitering to both interfaces and able to
communicate.  Is this possible?  What suggestions do you have?


  +-+
  | |
internal  | | external
--+ +-
192.168.1 | | real IP
  | |
  +-+
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Re: [Asterisk-Users] Voicemail with NFS

[Bruce Ferrell]

you might want to try autofs to drive the nfs functions.  it'll make you 
less susceptable as the filesystem won't be mounted full time


Douglas Garstang wrote:

I have /var/spool/asterisk/voicemail NFS mounted from another server. 
Everything is fine, until I simulate an NFS server failure, by shutting down 
the NFS server process.

At this point, Asterisk becomes almost non-responsive. It won't even process a 
'sip show peers' command correctly. It displays a few lines of text, pauses for 
several seconds, and then displays the rest. When a call comes into the system, 
Asterisk seems to do nothing for several seconds, and generally acts really 
sluggish. The phone gives up after several seconds, because Asterisk isn't 
doing anything.

I have used the soft option with the NFS mount.

I hope someone isn't going to tell me that the voicemail directory going away 
is going to cause Asterisk to fall in a heap on the floor. We just changed our 
model from a single, central voicemail server, to a distributed model, to get 
around some issues. We can't lose ALL pbx functionality just because the 
voicemail NFS server goes away. That's insane.

Doug.




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[Asterisk-Users] Asterisk on BSD?

[Bruce Ferrell]

The subject says it all I think.  I'm looking at maybe needing to run it 
under BSD 5


Thanks in advance

Bruce
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