[asterisk-users] How to change SIP header TO: ?

[Paul Mancheno H.]

Hello friends.

I have a softswitch in which I cannot create a list of blocked source
numbers; So, I have thought to use Asterisk and return a 302 message
when the number can make the call, my dialplan is as follows:

[from-external]
  exten => _AX.,1,Verbose(===> ${CALLERID(num)} to ${EXTEN})
   same =>  n,Set(MYDESTINY=${REPLACE(${EXTEN},A,)})
   same =>  n,Set(MYORIGIN=${CALLERID(num)})
   same =>  n,Set(ITEXISTS=${ODBC_GETBLACKPHONE(${MYORIGIN})})
   same =>  n,GotoIf($[${ITEXISTS}>0]?black)
   ; I will try to change header TO:
   same =>  n,Set(MYHEADER=${PJSIP_HEADER(read,To)})
   same =>  n,Set(MYnewHEADER=${REPLACE(MYHEADER,A,)})
   same =>  n,Set(PJSIP_HEADER(update,To)=${MYnewHEADER})
   ; The previous block did not work because the INVITE message is not
sent altered
   same =>  n,Transfer(PJSIP/sip:B${MYDESTINY}@10.1.1.2)
   same =>  n,NoOp(Transferencia=${TRANSFERSTATUS})
   same =>  n,Goto(end)
   same =>  n(black),Verbose(Fraudulento)
   same =>  n,Answer()
   same =>  n,Playback(bye)
   same =>  n,HangUp()
   same =>  n(end),Verbose(Terminado)

What I need is to be able to change the TO: header so that the
softswitch receives the number without the prefix "A" because even
though the call is completed efficiently I am having trouble with the
billing (which belongs to the softswitch).

Please can you help me and tell me how I can change the header TO: of
the message "302" (sent by the Transfer method).

Cheers,
Pepo.



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Re: [asterisk-users] asterisk mysql contacts

[Paul Neuwirth]

On Wed, 17 Jan 2018 09:26:28 -0700
John Kiniston  wrote:

> use func_odbc, create a new function that does a lookup.
> 
> [CALLERID]
> prefix=LOOKUP
> dsn=MyDB
> readsql=SELECT CALLERID from MyNames where CallerIdNum =
> '${SQL_ESC(${ARG1})}'
> 
> exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${CALLERID(NUM)}))
> 
> 
> On Wed, Jan 17, 2018 at 6:16 AM, Atux Atux  wrote:
> 
> > Hi. i have an asterisk 11 installation that i run in my soho
> > environment. My system has mysql to store all the cdrs.
> > I would like make use of the mysql and store numbers and names. eg
> > +4922123456789 "Atux Null". So when the +4922123456789 calls in my
> > system the name "Atux Null" will pop up next to the number.
> > at the moment i have a database called MyNames in mysql that has
> > this information, but i do not know how to make the dialplan read
> > from this database.
> > I would like to ask if there is a way to implement this easily in my
> > dialplan, please.
> >
> >

that is a good approach.. thinking over implementing this too.. and
filling the numbers using a trigger on cdr table.. 
is there a good manual how to set up odbc connection to mysql?  I
actually never used odbc on non microscrap systems..

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Re: [asterisk-users] queue peridiodic-announce-frequency

[Paul Neuwirth]

On Wed, 17 Jan 2018 12:08:40 +0100
Antony Stone <antony.st...@asterisk.open.source.it> wrote:

> On Wednesday 17 January 2018 at 11:59:21, Paul Neuwirth wrote:
> 
> > Hello group,
> > 
> > I tried a lot to enlarge the frequency (i.e. more announces, low
> > wait between). according to config, every 30 seconds the
> > announcement should take place. In fact, the first periodic
> > announce is done after 2 minutes?
> > What is my fault?  
> 
> Config snipped for clarity...
> 
> > [defaultq]
> > timeout = 10
> > retry = 99
> > wrapuptime=15
> > maxlen = 0
> > announce-frequency = 30
> > min-announce-frequency = 30
> > periodic-announce-frequency = 30
> > announce-holdtime = yes
> > announce-position = yes
> > announce-to-first-user = yes
> > periodic-announce = tt-allbusy,hold-or-dial-0
> > reportholdtime = yes  
> 
> I believe that both timeout and retry have to be less than the
> announce intervals.
> 
> In your case, retry is set to 99, with the announce intervals set at
> 30, therefore you would get announcements at 30, 60, 90, 120, 150...
> except that retry=99 stops any announcement before 99, therefore the
> first one you get is at 120 (2 minutes).
> 
> So, try setting retry to a value not less than 30 and see if that
> fixes things.
> 
> 
> 
> Antony.
> 

thank you. working fine now. I misinterpret the retry (thought counts).
Unfortunately the documentation (comments) lack dimensions almost at
all..

Regards

Paul

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[asterisk-users] queue peridiodic-announce-frequency

[Paul Neuwirth]

Hello group,

I tried a lot to enlarge the frequency (i.e. more announces, low wait
between). according to config, every 30 seconds the announcement should
take place. In fact, the first periodic announce is done after 2
minutes?
What is my fault?

Thank you

Regards 

Paul


# zypper if asterisk
Loading repository data...
Reading installed packages...


Information for package asterisk:
-
Repository : openSUSE_networkTelephony 
Name   : asterisk  
Version: 14.7.5-1.2
Arch   : x86_64
Vendor : obs://build.opensuse.org/network:telephony
Installed Size : 19.9 MiB  
Installed  : Yes   
Status : up-to-date
Source package : asterisk-14.7.5-1.2.src   
Summary: The Asterisk Open Source PBX


# grep -vE '^\s*;.*|^\s*$' queues.conf 
[general]
persistentmembers = yes
monitor-type = MixMonitor
[defaultq]
musicclass = default_noannounce
announce = tt-busy
strategy = ringall
context = qoutcon
timeout = 10 
retry = 99
wrapuptime=15
maxlen = 0
setinterfacevar=yes
setqueueentryvar=yes
setqueuevar=yes
announce-frequency = 30
min-announce-frequency = 30
periodic-announce-frequency = 30
announce-holdtime = yes
announce-position = yes
announce-to-first-user = yes 
periodic-announce = tt-allbusy,hold-or-dial-0
monitor-format = wav49 monitor-type = MixMonitor
reportholdtime = yes
ringinuse = yes 
member => SIP/XXW
member => SIP/XXX
member => SIP/XXY
member => SIP/XXZ
member => SIP/XXA

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Re: [asterisk-users] remote Asterisk console

[Paul Neuwirth]

On Tue, 16 Jan 2018 18:18:18 +0200
Tzafrir Cohen <tzafrir.co...@xorcom.com> wrote:

> On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote:
> > Hello group,
> > 
> > what is the preferred method to connect to asterisk cli over
> > network? I need to run asterisk cli commands remotely.  
> 
> As others have mentioned: the manager interface is normally better for
> running over network.
> 
> The manager interface also has an action calld 'Command' that runs a
> CLI command. In fact, contrib/scripts/astcli uses it to allow
> providing a remote console.
> 
> Permissions needed for your manager user: For most things just:
> 
> write=command
> 
> To also be able to originate calls:
> 
> write=command,originate
> 
> To also be able to restart / reload:
> 
> write=command,system
> 
> > Sharing the unix socket through NFS, if that's working?  
> 
> No.
> 
> > Or any other approaches, despite using SSH or rlogin, rsh.  
> 
> SSH: should work, sure. However, it means you ssh to root at the
> remote host. Better set a key with 'command' explicitly set in
> authorized_keys for this.
> 
> Rlogin, rsh: seriously? Anybody still uses those? Not only are they
> way less secure than SSH, they are also way less conveninet than any
> decent SSH implementation.
> 
> Anyway, as mentioned before: you should probably use AMI.
> 

Thank you both. That was (most likely) what I was looking for - but
still some worries about sending plaintext passwords... For my simple
commands a simple netcat command works for me. Previously used asterisk
-rx in scripts. But now asterisk servers and other processes are split
over multiple physical servers.
A binary or script, making use of encryption and miming asterisk -r
would be best. I am wondering, why such a tool is not part of asterisk
itself... maybe I give this a try setting up a user (group asterisk)
with asterisk -r as "login shell".. and use ssh.. or something like
that. It should be that safe, no other commands can be executed..

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[asterisk-users] remote Asterisk console

[Paul Neuwirth]

Hello group,

what is the preferred method to connect to asterisk cli over network? I
need to run asterisk cli commands remotely.
Sharing the unix socket through NFS, if that's working?
Or any other approaches, despite using SSH or rlogin, rsh.

Thank you

Paul

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Re: [asterisk-users] load-balancing AMI and load-balancing FastAGI?

[Paul Simon]

Anyone?
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[asterisk-users] load-balancing AMI and load-balancing FastAGI?

[Paul Simon]

Hi,

I am starting a new project to develop a predictive dialler system.

- Agents can start receiving calls from the queue if agent press
Available button on the browser which will unpause the queue on Asterisk.

- About 100-150 concurrents calls on a Asterisk box

- Call-out initiated. Other end answers. Passes AMD. Lands in Queue and
direct to agents that is available and call is recorded.

- Update state of the call (Ringing, Talking, etc) on the database.

- Listen the events such as Hang Up from customer, check if call is
successfully originated or what the failure, etc.

- Agent will have ability to transfer customer call to other agent or
external number.
As described above to develop a predictive dialler system, is it best to
use AMI or FastAGI?

I am aware that I can setup FastAGI load balancing such as agitator
(FastAGI reverse proxy).

AMI case: load-balances incoming events/response across multiple processes
(multiple AMI connections on the same asterisk machine), should the
ami events/response should be pushed into RabbitMQ so the proess can read
from RabbitMQ ?

Thanks
Paul
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Re: [asterisk-users] small homebrew pbx

[Paul Hayes]

On 15/06/15 07:46, lu...@sulweb.org wrote:

Hello all,


Given the requirements above, what's a cheap but working PCIe card / USB
adapter I could buy for this kind of PBX? Do I need things like echo
cancellation? Do I need FXS ports?

Thanks in advance,
Lucio.



I would get hold of some lower-power hardware, that system seems hugely 
over-specified for what you want to do.


A raspberry pi  a Cisco SPA-3102 would be a good solution.  Cisco don't 
make the 3102 any more but there are still plenty of them around.  I 
believe Grandstream still make ATAs as well but I've never thought very 
highly of them.  As others have said, it's an FXO port you need.


You want to avoid transcoding on low power hardware such as a raspberry 
pi so set everything for a codec such as g711a or g711u (Asterisk, the 
IP phones you use and the SPA3102).



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Re: [asterisk-users] json.c:704 ast_json_vpack: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.

[Paul Belanger]

On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov d...@belkam.com wrote:
 Hello!

 Just installed asterisk 13.2.0 and see many such messages in log, I see them
 in console during calls, really something like this:


 -- Executing [6166@kanbaikal:2] Dial(OOH323/kanbaikal-6,
 SIP/6166@asterisk) in new stack
   == Using SIP RTP TOS bits 184
   == Using SIP RTP CoS mark 5
 -- Called SIP/6166@asterisk
 0x7fa9d4007660 -- Probation passed - setting RTP source address to
 10.18.0.19:26052
 -- SIP/asterisk-000c is making progress passing it to
 OOH323/kanbaikal-6
 -- SIP/asterisk-000c is ringing
 0x7fa9d4007660 -- Probation passed - setting RTP source address to
 10.18.0.19:26052
 0x7fa9d800d610 -- Probation passed - setting RTP source address to
 192.168.166.2:2080
 [Mar  5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 [Mar  5 11:13:14] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 -- SIP/asterisk-000c answered OOH323/kanbaikal-6
 -- Channel OOH323/kanbaikal-6 joined 'simple_bridge' basic-bridge
 d751932e-2e26-4671-8fcc-048b20156ec9
 -- Channel SIP/asterisk-000c joined 'simple_bridge' basic-bridge
 d751932e-2e26-4671-8fcc-048b20156ec9
 0x7fa9d800d610 -- Probation passed - setting RTP source address to
 192.168.166.55:3098
 [Mar  5 11:13:19] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 [Mar  5 11:13:19] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 [Mar  5 11:13:24] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 [Mar  5 11:13:24] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.
 [Mar  5 11:13:29] ERROR[3526]: json.c:704 ast_json_vpack: Error building
 JSON from '{s: s, s: s}': Invalid UTF-8 string.


 This is call from H323, as I know avaya ,  chan_ooh323 from my side to
 another asterisk SIP chan_sip on both sides.

 Just because everything work OK, I , definitely, can comment out this error
 message, but...
 Could you give me any idea why this error can appear?

If you haven't create an issue on Jira, this is a bug.

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Re: [asterisk-users] WebRTC phone

[Paul Belanger]

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens jar...@mogl.com wrote:
 For those that were interested I have attached the kamailio.cfg which we
 have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the
 following yum packages:

 kamailio.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-auth-ephemeral.x86_64   4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-bdb.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-carrierroute.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-cpl.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-debuginfo.x86_644.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-gzcompress.x86_64   4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-ims.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-ldap.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-mysql.x86_644.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-outbound.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-postgres.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-presence.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-python.x86_64   4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-sctp.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-snmpstats.x86_644.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-sqlite.x86_64   4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-tls.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-unixodbc.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-utils.x86_644.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-uuid.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-websocket.x86_644.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-xml.x86_64  4.2.1-4.1
 @home_kamailio_v4.2.x-rpms
 kamailio-xmpp.x86_64 4.2.1-4.1
 @home_kamailio_v4.2.x-rpms

 Keep in mind that using Kamailio to bridge the signalling is only half of
 the equation. You must also bridge the media and so the rtpengine module
 allows Kamailio to interface with the rtpengine
 (https://github.com/sipwise/rtpengine) which does that half.

 In the provided example Kamailio.cfg there isn't any real hardening and it's
 pretty much purely used as a bridge that would front an Asterisk 1.8 server
 for webrtc capabilities (but not any other sip). It uses the dispatcher
 module to dispatch to the underlying asterisk so you will still need to add
 the Asterisk to the dispatcher config.

+1 to everything here. We also do this and it works quiet well.  Kudos.

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Re: [asterisk-users] Is Asterisk a Linux only system?

[Paul Belanger]

On Thu, Feb 12, 2015 at 11:40 AM, Matthew Jordan mjor...@digium.com wrote:


 On Thu, Feb 12, 2015 at 8:52 AM, D'Arcy J.M. Cain da...@vex.net wrote:

 On Thu, 12 Feb 2015 09:43:33 -0500
 Ron Wheeler rwhee...@artifact-software.com wrote:
  Why not just bite the bullet and move to a supported Linux?

 If all I had was a phone switch that might be an option but this is
 just part of a multi-server system that needs to be able to move
 services back and forth so the underlying OS has to be the same for
 everything.  Besides, I am a NetBSD developer and so I am also
 interested in making every package rock solid on it.

  - you can be assured that it works
  - updates are tested

 I would be willing to make a NetBSD machine (not my production server)
 available for running unit tests.  Are there already unit tests in the
 distribution?


 Yes there are. In addition to unit tests, there are also the functional
 tests in the Asterisk Test Suite [1].

 To enable them as well as set up Asterisk for the Test Suite:

 1. Configure Asterisk for development mode:
   $ ./configure --enable-dev-mode
 2. In menuselect, enable the TEST_FRAMEWORK Compiler Flag
 3. Also in menuselect, enable the Test Modules. These provide the unit
 tests.
 4. Build/install Asterisk
 5. Run Asterisk
 6. Execute the unit tests (or a subset thereof) using the CLI:
   *CLI test execute [category|all]

 Note that some unit tests require a particular configuration or certain
 subsystems to be enabled. You can examine the CI build agent scripts used
 for test runs here:

 http://svn.asterisk.org/svn/testsuite/bamboo/trunk/bin/

 Specifically, the build-asterisk-only.sh script and
 run-asterisk-unittests.sh.

 Setting up [2] and running [3] the Asterisk Test Suite is documented on the
 wiki, and generally covers a lot more functionality than the unit tests.

 [1]
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation
 [2]
 https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suite
 [3]
 https://wiki.asterisk.org/wiki/display/AST/Running+the+Asterisk+Test+Suite

It should be noted, we did have a FreeBSD and Ubuntu systems running
the testsuite back in 2010.  FreeBSD was donated to the project.

I personally had a PowerPC system running asterisk / testsuite, on debian.

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Re: [asterisk-users] Question regarding custom announcements used by several Asterisk servers

[Paul Belanger]

On Fri, Feb 6, 2015 at 5:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:


 On 6 February 2015 at 07:54, Olli Heiskanen ohjelmistoarkkite...@gmail.com
 wrote:


 Hello,

 Got a question regarding custom announcements in Asterisk.

 My goal is to allow my users record their own queue announcements and
 choose which announcements they want to use in each queue. I have several
 Asterisk servers and a Kamailio server which dispatches call traffic between
 the Asterisks. Question is, is it possible to have something like a NSF disk
 shared between several asterisk servers and store custom announcements
 there, where all Asterisks would use them? I expect to have to place the
 files under whatever I configure in asterisk.conf. Additionally, can I place
 the announcements in subfolders under that directory and in my realtime
 queue table use values something like '/subfldr/myannouncement'?

 Keep up the good work!

 cheers,
 Olli

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 Hi

 All of that is possible and is exactly what we do, both for customer sounds
 and for call recordings. Just make sure you have resilience in your shared
 storage device.

 Alternatively, you could use something like Puppet to deploy the files to
 all the servers.

This is basically what we do, we use puppet to help distribute files
to remote servers while still using app_queue.  Shared network drive
also works.

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Re: [asterisk-users] Asterisk Java API - Up to date

[Paul Belanger]

On Tue, Jan 27, 2015 at 4:14 PM, symack sym...@gmail.com wrote:
 Hello Everyone,

 I am required to write a java program that will get our asterisk to:

 * Query the database for phone numbers
 * Loop through numbers and dial
 * Play message
 * Get dial pressed response
   - If 1 = Yes
   - If 2 = No
   - If 3 = Connect to Agent
 * AMD Capable
 * Disposition

 I am proficient with Java and found the Asterisk-Java API. My questions
 are:

 * What is the recommended API to use
 * Is Asterisk-Java API maintained by digium
 * Am I overlooking anything?

 Your help is greatly appreciated.

There's many ways to accomplish this, many have been discussed on this
mailing list.

You are going to use the AMI to originate calls into asterisk.  No,
Asterisk-Java is not maintained my Digium.  As for overlooking,
likely, but you should be able to see anything you missed in your
testing phase.

You should be able to google Asterisk dialers to see some example that
people have done.

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Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

[Paul Belanger]

On Wed, Jan 28, 2015 at 1:37 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
 On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi

 We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
 queues.

 For a particular customer, when I run queue show queue_name I get the
 following numbers:

 queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s
 holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s

 So from that data we look at
 17s holdtime
 And assume that is the average hold time before calls get answered by a
 queue members.

 However, if I calculate the average hold time from out queue log table using
 the following SQL

 select sum(data1)/ count(*) as ave_hold_time from queue_log where time 
 DATE(NOW()) and queuename='queue_name' and event='CONNECT';

 I get the vastly different figure of 92.4.

 So, is the queue show figure wrong due to a bug or am I making an incorrect
 assumption as to what it means?

 Thanks in advance

 Welcome to business logic embedded into app_queue.  The issue with the
 queue show command rendering stats, is what timeframe are the stats
 aggregated over?  IIRC, the calculations are using a moving
 average[1].

Opps, sent instead of pasting.

Either way, your likely better off rendering the data using the raw
sql info vs depending on CLI output.  That's what we've done.

[1] http://en.wikipedia.org/wiki/Moving_average
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Re: [asterisk-users] queue show queue-name vs queue log for calculating average hold time

[Paul Belanger]

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi

 We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for
 queues.

 For a particular customer, when I run queue show queue_name I get the
 following numbers:

 queue_name has 0 calls (max unlimited) in 'ringall' strategy (17s
 holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s

 So from that data we look at
 17s holdtime
 And assume that is the average hold time before calls get answered by a
 queue members.

 However, if I calculate the average hold time from out queue log table using
 the following SQL

 select sum(data1)/ count(*) as ave_hold_time from queue_log where time 
 DATE(NOW()) and queuename='queue_name' and event='CONNECT';

 I get the vastly different figure of 92.4.

 So, is the queue show figure wrong due to a bug or am I making an incorrect
 assumption as to what it means?

 Thanks in advance

Welcome to business logic embedded into app_queue.  The issue with the
queue show command rendering stats, is what timeframe are the stats
aggregated over?  IIRC, the calculations are using a moving
average[1].




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Re: [asterisk-users] Cannot get my first WebRTC experiment to work.

[Paul Belanger]

On Wed, Jan 28, 2015 at 8:27 AM, Antonio Gómez Soto
antonio.gomez.s...@gmail.com wrote:
 Hi all,

 Trying to do my first WebRTC. Using stock asterisk 1.13.0.
 I setup the asterisk according to the recipe on the wiki, but cannot get it
 to work.
 Dialing from sipml5 on chrome I get no sound, regular bria on standard sip
 works.

 My network setup by the way: I am working from a cable modem, I created the
 test setup at digital ocean. From my laptop I also have a direct VPN
 connection
 to the asterisk server my laptop being 192.168.241.10 and asterisk being
 192.168.241.30

 I think something is wrong with the RTP address negotiation, but I have
 trouble
 interpreting the SDP wrt WebRTC and ICE.

 1. asterisk seems to be telling sipml5 to send audio to it's public ip
 addres, but * sends to 192.168.241.10
 2. the asterisk output does show RTP flows to chrome, but there's no sound
 from chrome.

 I hope someone can intersperse the output with comments?

Pastebin the fill debug, you've delete an important piece of information.

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Re: [asterisk-users] ${HASH(SIP_CAUSE,channel-name)}

[Paul Belanger]

On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
 Hello,

 I read on the wiki :

 Asterisk 1.8 will allow to read SIP response codes in the dialplan via
 ${HASH(SIP_CAUSE,channel-name)}. Additionally make sure you're using the
 destination channel, not the source channel.

 But when I use this in my dialplan, this 'variable' is empty.

 Dialplan :

 exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
 exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})

 CLI :

 [Oct 30 14:48:03] -- Executing [h@pbx-routing:5]
 NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
 [Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
 NoOp(SIP/SipAT01-0015, sip cause = ) in new stack


 Can anyone tell me how this should be used ?

sip.conf: storesipcause=yes


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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

[Paul Albrecht]

On Oct 29, 2014, at 2:45 PM, Ben Klang bkl...@mojolingo.com wrote:

 
 On 10/28/2014 06:03 PM, Ben Langfeld wrote:
 On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote:
 What is the alternative to the dial plan? Is everyone talking about getting 
 rid of the statements like:
 exten = s,1,
 
 what is the alternative? 
 
 Remote applications based on APIs like ARI. This is the start of the 
 discussion, and please remember that nothing has been decided or even 
 presented as a robust plan yet. This is brain-storming.
 
 Additionally, note that the original proposal was to deprecate AMI/AGI in 
 favour of ARI once it is feature complete with those protocols; an entirely 
 lesser change than the removal of the dialplan in its entirety.
  
 
 Since this thread has my name on it, I guess it’s past time that I explain my 
 motivation for making the suggestion, and try to restore some of the context 
 that was present in the discussion at AstriDevCon.
 
 Before I jump into the details of my proposal, I’d like to clarify terms...
 

It’s intellectually dishonest to redefine the terms of an argument to 
presuppose your own conclusion. If you don’t intend to use the term “deprecate” 
as it is commonly understood by software developers and users than you should 
avoid the use of the term “deprecate” so that others clearly understand your 
argument. If you really mean “deprecate” as commonly understood by software 
developers and users then you should be prepared to defend that proposition.
 
 Now, on to what I originally proposed...
 

It’s clear from the title of the agenda item what was proposed. You proposed 
deprecating AMI/AGI and that entails deprecating the dial plan. The fact that 
deprecating the dial plan is now on the table is a direct consequence of your 
proposal. This is reflected in both comments made at AstiCon and Matt’s summary 
of  Astricon on the development list. You can’t have it both ways. You want to 
deprecate dial plan or not. Which is it? 

 It is my opinion that while AGI and AMI are probably individually fixable, 
 doing so would cause backward-incompatible changes…

Deprecating the dial plan and AGI/AMI is incompatible going forward. What is 
supposed to happen? Are users supposed to throw away there applications 
whenever ARI/Stasis is feature complete? Is ARI/Stasis really any easier to use 
than the dial plan? Are we all supposed to use Adhearsion? 



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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 29, 2014, at 4:26 PM, Matthew Jordan mjor...@digium.com wrote:

 On Wed, Oct 29, 2014 at 2:31 PM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 28, 2014, at 5:03 PM, Ben Langfeld b...@langfeld.me wrote:
 
 On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote:
 
 What is the alternative to the dial plan? Is everyone talking about
 getting rid of the statements like:
 exten = s,1,
 
 what is the alternative?
 
 
 Remote applications based on APIs like ARI. This is the start of the
 discussion, and please remember that nothing has been decided or even
 presented as a robust plan yet. This is brain-storming.
 
 
 We’re not at the start of the “discussion” to deprecate the dial plan. The
 start of the “discussion” began when some developers decided to try standing
 Asterisk on its head by adding  “asynchronous AGI.” Evidently, that was good
 so then they continued the “discussion” by adding ARI/Stasis. Now the
 “discussion” is in full career as ARI/Stasis has metastasized beyond its
 original scope to encompass all of Asterisk. None of said “discussion” ever
 happened on the lists nor was the broader Asterisk community involved as far
 as I can determine. A parallel “discussion” was started by a shill at
 AstiCon this year to begin to get the “vast unwashed” onboard with
 ARI/Stasis, that is, so that Matt could come back from AstiCon claiming that
 the broader Asterisk community is in agreement that ARI/Stasis is the future
 of Asterisk and that the dial plan can be deprecated. The inevitable result
 of these parallel paths is a completely predictable train wreck when the
 developers designing features that users don’t want crash into users who
 have been using Asterisk as originally designed.
 
 Additionally, note that the original proposal was to deprecate AMI/AGI in
 favour of ARI once it is feature complete with those protocols; an entirely
 lesser change than the removal of the dialplan in its entirety.
 
 
 So you're saying that deprecating the dial plan is not on the table? How
 then do you explain statements like this: Leif: we're in a transition,
 moving from dialplan model to external control model.  Probably need
 external application to be built for us to move completely away from
 AMI/AGI.” or  this Paul: take away apps, and whatever is in the core is
 what we should care about.”
 
 
 Paul:
 
 This is a notice that you are in violation of the Asterisk community
 code of conduct:
 
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Community+Code+of+Conduct
 
 You have repeatedly insulted members of the Asterisk community using
 derogatory language that is inappropriate for this mailing list. This
 creates a hostile atmosphere that makes it difficult for productive
 communication to occur, which is the lifeblood of this open source
 project. Members of an open source community should not feel like they
 are under attack merely for expressing an opinion. While we value the
 opinions you bring to the discussion, your tone and choice of language
 is completely inappropriate and will not be tolerated.
 
 If you continue to use inflammatory language and rhetoric, you will be
 banned from participation in the Asterisk project.
 
 Matt
 
 -- 
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org


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Re: [asterisk-users] Asterisk 12 - zombie processes

[Paul Belanger]

On Tue, Oct 28, 2014 at 11:10 AM, Yaron Nachum nachum.ya...@gmail.com wrote:
 Mathew,
 When I run 'ps -ef|grep asterisk' the following processes are displayed:
 root  6861 1  0 Aug27 ?00:00:00 /bin/sh
 /ivr/app/asterisk/sbin/safe_asterisk -U asterisk -G asterisk -C
 /ivr/app/asterisk/etc/asterisk/asterisk.conf
 asterisk  8062  6861  3 Oct27 ?00:44:56
 /ivr/app/asterisk/sbin/asterisk -f -U asterisk -G asterisk -C
 /ivr/app/asterisk/etc/asterisk/asterisk.conf -vvvg -c
 root 20776  2200  0 11:20 pts/200:00:33 tail -f asterisk.log
 asterisk 23076  8062  0 17:01 ?00:00:00 [asterisk] defunct
 asterisk 23897  8062  0 17:03 ?00:00:00 [asterisk] defunct

 also when I run top the same amount of zombie processes are displayed:
 Tasks: 185 total,   1 running, 182 sleeping,   0 stopped,   2 zombie

 Regarding the AGI - we are using AGI in order to run php scripts for
 external logic. I have printed the PIDs of the php scripts and none of them
 are related to the PID's of those zombie processes.
 Do you have any idea how to find out what are these processes?
 Yaron.

Are you doing anything like:

# asterisk -rx 'core show channels'

via an external process?

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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

[Paul Albrecht]

On Oct 28, 2014, at 5:03 PM, Ben Langfeld b...@langfeld.me wrote:

 On 28 October 2014 19:47, Derek Andrew derek.and...@usask.ca wrote:
 What is the alternative to the dial plan? Is everyone talking about getting 
 rid of the statements like:
 exten = s,1,
 
 what is the alternative? 
 
 Remote applications based on APIs like ARI. This is the start of the 
 discussion, and please remember that nothing has been decided or even 
 presented as a robust plan yet. This is brain-storming.
 

We’re not at the start of the “discussion” to deprecate the dial plan. The 
start of the “discussion” began when some developers decided to try standing 
Asterisk on its head by adding  “asynchronous AGI.” Evidently, that was good so 
then they continued the “discussion” by adding ARI/Stasis. Now the “discussion” 
is in full career as ARI/Stasis has metastasized beyond its original scope to 
encompass all of Asterisk. None of said “discussion” ever happened on the lists 
nor was the broader Asterisk community involved as far as I can determine. A 
parallel “discussion” was started by a shill at AstiCon this year to begin to 
get the “vast unwashed” onboard with ARI/Stasis, that is, so that Matt could 
come back from AstiCon claiming that the broader Asterisk community is in 
agreement that ARI/Stasis is the future of Asterisk and that the dial plan can 
be deprecated. The inevitable result of these parallel paths is a completely 
predictable train wreck when the developers designing features that users don’t 
want crash into users who have been using Asterisk as originally designed.

 Additionally, note that the original proposal was to deprecate AMI/AGI in 
 favour of ARI once it is feature complete with those protocols; an entirely 
 lesser change than the removal of the dialplan in its entirety.

So you're saying that deprecating the dial plan is not on the table? How then 
do you explain statements like this: Leif: we're in a transition, moving from 
dialplan model to external control model.  Probably need external application 
to be built for us to move completely away from AMI/AGI.” or  this Paul: take 
away apps, and whatever is in the core is what we should care about.”

  
 
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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

[Paul Albrecht]

Not asking for unqualified promises about the future of Asterisk. Simply asking 
for an acknowledgment of the obvious, that is, Asterisk without the dial plan 
wouldn’t be Asterisk. The fact that one is not forthcoming raises a red flag 
with respect to the future of Asterisk. Furthermore, adding “asynchronous AGI”  
and ARI/Stasis to Asterisk is similarly a cause for concern since it’s a 
complete break with the original Asterisk design. Since Asterisk is an open 
source community supported project, one would expect the consultants/developers 
pushing these changes would be willing to share their vision with the rest of 
the Asterisk community.

On Oct 27, 2014, at 2:32 PM, Jeffrey Ollie j...@ocjtech.us wrote:

 On Mon, Oct 27, 2014 at 2:04 PM, Paul Albrecht palbre...@glccom.com wrote:
 
 The reason the dial plan can never be deprecated is because Asterisk 
 wouldn’t be Asterisk without the dial plan. Sure, you could re-engineer 
 Asterisk so that it would be “better for a small select group of users at 
 the expense of the majority of community that use the product as designed 
 for the purpose it was originally intended. However, you’re either very 
 naive or delusional if you think the community is going to follow you down 
 that path. Do you really believe the community is going simply chuck their 
 dial plans and walk away from their investment in Asterisk? Not likely, dude.
 
 My comment/question wasn't really about dial plans, per se.  My
 question was about you insisting that Digium make such unqualified
 promises about the future of Asterisk.  Even though Digium is a
 private company, I believe that they are still bound by U.S. laws
 regarding forward-looking statements[1].
 
 So even if they wanted to (which I doubt), there's no way you're going
 to get the promise that you're looking for.
 
 [1] http://en.wikipedia.org/wiki/Forward-looking_statement
 
 -- 
 Jeff Ollie
 
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[asterisk-users] AppKonference 2.6

[Paul Albrecht]

I have released an updated AppKonference that compiles with Asterisk 13. You 
can download the latest code from source forge: 
sourceforge.net/projects/appkonference

That said Asterisk 13 doesn’t get that much attention because I use Asterisk 
1.4 + some hacks. Here’s a link to my Asterisk 1.4 github repository: 
https://github.com/pjalbrecht/asterisk You don’t need these hacks to use the 
module, but you may find them useful.-- 
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Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI (Ben Klang)

[Paul Albrecht]

The reason the dial plan can never be deprecated is because Asterisk wouldn’t 
be Asterisk without the dial plan. Sure, you could re-engineer Asterisk so that 
it would be “better for a small select group of users at the expense of the 
majority of community that use the product as designed for the purpose it was 
originally intended. However, you’re either very naive or delusional if you 
think the community is going to follow you down that path. Do you really 
believe the community is going simply chuck their dial plans and walk away from 
their investment in Asterisk? Not likely, dude. 

On Oct 24, 2014, at 11:48 AM, Jeffrey Ollie j...@ocjtech.us wrote:

 On Fri, Oct 24, 2014 at 10:09 AM, Paul Albrecht palbre...@glccom.com wrote:
 
 When Matt says deprecating the dial plan would be difficult and would take a
 long time it seems to me he’s being evasive and misleading. He doesn’t say
 it’s never going to happen and he doesn’t share whatever he thinks the
 Asterisk vision actually is which he should presumably be aware of since he
 is the Asterisk engineering manager.
 
 Why do you keep insisting that Digium promise to *never* deprecate
 dial plans?  I don't think that's a promise that's really worth
 anything as there may be really good reasons in the future to do so.
 I think that you've gotten the best that you will get: they've said
 that there are no plans within Digium to deprecate the dial plan, and
 if there were plans, they'd give people a long time prepare before it
 actually happens.
 
 It's probably a good time to refresh your understanding of Digium's
 support policies:
 
 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
 
 Version 13 will be around until at least 2018, so you'll have *at
 least* that long to prepare for the switch, since version 13 is
 feature frozen so there's no way the dial plan would be removed from
 13.
 
 And all of this talk of deprecating the dial plan isn't even coming
 from Digium.  It's something that was suggested by a community member
 at the developer conference.  I wasn't there so I don't know how
 seriously it was taken there, but it would have been impolite of
 everyone involved to just ignore it.
 
 -- 
 Jeff Ollie
 
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Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 23, 2014, at 1:58 PM, Kevin Larsen kevin.lar...@pioneerballoon.com 
wrote:

  From: Paul Albrecht palbre...@glccom.com 
 
  Seems like now is as good a time as any to raise these issues, in 
  fact, sooner is better than later because once developers start down
  a path it’s very difficult to get them change their minds no matter 
  how much sense it makes. The fact that developers are even 
  considering taking away user functionality like the dial plan is in 
  of itself a very serious problem because it demonstrates they don’t 
  see Asterisk from the user perspective. 
 
  Don’t object to extending the Asterisk user interface or changing 
  Asterisk internals. Do object to is taking away functionality that 
  users expect, are familiar with, and has made the Asterisk project 
  successful. 
 
  Then your experience is atypical. Asterisk has been unstable for 
  several years as developers have continually shoveled new features 
  into the code base over several releases. That’s not necessary 
  objectionable, it’s even to be expected; however, at some point 
  developers need to turn their attention to less glamorous less 
  exciting things like stability and performance. 
 
 I don't think anyone is objecting to you bringing this up, as it has been 
 mentioned at the dev con. Perhaps it is just that the tone doesn't come 
 across properly in an email, but you are coming across as confrontational and 
 alarmist and it seems to be setting people on edge. Matt has already chimed 
 in that he doesn't see how it would be possible to deprecate the dial plan at 
 this time and even if it were possible, the process would take on the order 
 of years, giving you plenty of time to enact any contingency plans you might 
 need. Scott G. from Digium even posited that if it were to be removed from 
 the core, it would likely end up as a loadable module so that it wouldn't 
 burden those who don't need it and could be loaded for those who do. 
 

When Matt says deprecating the dial plan would be difficult and would take a 
long time it seems to me he’s being evasive and misleading. He doesn’t say it’s 
never going to happen and he doesn’t share whatever he thinks the Asterisk 
vision actually is which he should presumably be aware of since he is the 
Asterisk engineering manager.

As for Scott’s suggestion, I don’t see how you can have it both ways: on the 
one hand ARI can’t work in an environment supporting AGI/AMI and the dial plan, 
and on the other you can support AGI/AMI in an optional module. It’s just not 
believable. If you’re not convinced, run menuconfig and look through the 
applications and functions sections. All of this stuff would have to change and 
you think that at the end of that process the dial plan would survive? I don’t 
think so.

 These developers do not exist in a vacuum, nor do they have total control 
 over where Asterisk goes. Influence, sure, but there is still a corporate 
 structure out there that finds it necessary to be customer oriented. They 
 would have to be monumentally stupid (something which I haven't seen previous 
 evidence of) to kill off the dial plan without providing a path forward for 
 those who depend on it. Furthermore, even if they did pull a stunt so bad as 
 to alienate half their users, the open source code would be forked so fast as 
 to make your head spin or people would migrate to other similar packages 
 (Freeswitch comes to mind). Digium sells their own PBX hardware that I am 
 sure depends on these technologies that you are afraid will go away. They 
 have direct skin in this game too. 
 

Totally get why Ben Klang thinks everyone should use adhearsion and that all 
the resources of the Asterisk community should be devoted to his project. But 
of course everyone thinks their project is the most important project in the 
world. What I don’t get is why one project or framework is deemed to be so 
important that is trumps everyone else in the Asterisk community.

 I would be interested to know just how atypical my experience is. I have 
 found that on my 1.6 systems I would have random crashes over time. After 
 upgrading over multiple sites, my 11.x systems have been rock solid for the 
 most part. I did have a case where I did a store and forward of a fax that if 
 I tried to forward the fax and it had no file to forward would cause a crash, 
 but other than that, I haven't seen any problems in normal day to day usage. 
 I always thought that the general consensus was that the 11.x series was 
 quite a bit more stable than the older versions. 
 

Wouldn’t use the 1.6 release as a basis of comparison as that release was 
regarded as DOA by most folks. A better example would be Asterisk 1.4. We use 
it because it’s stable, we don’t need any of the new features and definitely 
want to avoid the performance degradation in the later releases.

Having said that, Asterisk 1.6 was an interesting release because that’s when 
“async agi” was introduced

Re: [asterisk-users] AstriDevCon 2014:Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 3:27 PM, Kevin Larsen kevin.lar...@pioneerballoon.com 
wrote:

  From: Paul Albrecht palbre...@glccom.com 
  Here’s a link to the minutes:  https://wiki.asterisk.org/wiki/
  display/AST/AstriDevCon+2014 
  
  It has you saying: Leif: we're in a transition, moving from dialplan
  model to external control model.  Probably need external application
  to be built  for us to move completely away from AMI/AGI. 
  
  So you’re saying Asterisk is moving away from the dial plan or were 
  you misquoted? 
 
 Paul, 
 
 I think you are getting worked up way too early in this process. This is one 
 comment with only a little bit of context surrounding it. Such a major change 
 would take quite awhile to make and there would be plenty of warning before 
 it happens, with plenty of opportunities to discuss. The dial plan isn't 
 going away tomorrow and if it does ever go away, there will be plenty of time 
 to work out a transition plan. 
 

Seems like now is as good a time as any to raise these issues, in fact, sooner 
is better than later because once developers start down a path it’s very 
difficult to get them change their minds no matter how much sense it makes. The 
fact that developers are even considering taking away user functionality like 
the dial plan is in of itself a very serious problem because it demonstrates 
they don’t see Asterisk from the user perspective.

 Looking at the path development has taken, it seems pretty clear that they 
 have been working towards enabling greater external control of what Asterisk 
 does, making it the engine that can drive other media applications. Doesn't 
 mean it can't and won't be used as a traditional pbx, but to grow what it 
 does will require some changes. 
 
 If being a mature part of Asterisk means that something shouldn't be changed, 
 we should also protest the move from the current SIP stack to pjsip. There 
 are any number of reasons to deprecate mature code. It may not be needed or 
 something better may come along. 
 

Don’t object to extending the Asterisk user interface or changing Asterisk 
internals. Do object to is taking away functionality that users expect, are 
familiar with, and has made the Asterisk project successful.

 All I can say is that having experience with a few versions of Asterisk, it 
 seems to get better and more stable as new versions come along. Perhaps a bit 
 of faith that they are not trying to kill off their product simply by having 
 a discussion at a dev conference is in order. 
 

Then your experience is atypical. Asterisk has been unstable for several years 
as developers have continually shoveled new features into the code base over 
several releases. That’s not necessary objectionable, it’s even to be expected; 
however, at some point developers need to turn their attention to less 
glamorous less exciting things like stability and performance.

 Kevin Larsen -- 
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Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 3:39 PM, Matthew Jordan mjor...@digium.com wrote:

 
 On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote:
 
 
 On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
 
  Paul Albrecht wrote:
  Really? Shouldn’t something this major affecting the entire Asterisk
  community get discussed on the lists? Any idea what Leif is talking
  about when he says the community is in transition, moving from dial
  plan model to external control.
 
  It was something Ben Klang brought up and wanted to talk about - it's
  not something that has been decided 'nor does anyone know what the
  future entails. Any further discussions will naturally occur on the
  mailing list and in fact some things have explicit action items to bring
  them up on here.
 
 
 The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy 
 talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the 
 first place. It’s completely impractical and can never happen. Moreover, 
 Leif seems to think we (the asterisk community) are in transition. What does 
 that mean? Are we abandoning the dial plan? Seriously? That’s never gonna 
 happen either. ARI isn’t easier to use than dial plan scripting. I guess one 
 could hope that what happens in Vegas stays in Vegas”, but I don’t think 
 the Asterisk community has that kind of luck.
 
 
 Just because someone decided to bring up a radical idea does not mean we 
 refuse to discuss it.
 
 
 So you agree that deprecating AMI/AGI is “crazy talk” but you’ll discuss it 
 because of your open-mindedness?
 
 I didn't say that the idea of deprecating AMI/AGI is crazy talk. I did say 
 that radical ideas - and even ones that some folks think are crazy - are all 
 fine to discuss at AstriDevCon.
 
 The whole point of AstriDevCon is to have a large, free, and open 
 conversation about Asterisk Development. I fundamentally disagree with the 
 notion that that should be discouraged.
 

The problem with AstriConDev is there is no user input so what you have is a 
developer echo chamber and what you get is groupthink.

 This is an open source project. Communication is done in an open, 
 transparent manner. People should feel like they can bring up interesting, 
 radical, and yes - even crazy - ideas.
 
 
 By the same token, when you propose ideas, you must be prepared for honest 
 criticism and accept it in graciously rather than simply resorting to 
 argument ad hominem. 
 
 You didn't have honest criticism. You labelled a discussion point as crazy 
 talk and said we shouldn't have even discussed it.
 
 There was no ad hominem attack. I never attacked you. I never even attacked 
 your statements. I simply defended the free exchange of ideas in AstriDevCon. 
 I have no problem doing that.
 
 On the other hand, you did callously label an Asterisk Developer's admittedly 
 ambitious idea as crazy talk. In the future, you may want to choose your 
 language more carefully if you wish for others to have a more open discussion 
 with you.
 
  
 If you don't like that, you don't have to participate in the discussion.
 
 
 You haven’t really responded to the substance of my post, that is, is 
 asterisk abandoning the dial plan?
 
 There are Asterisk users (who also happen to develop) who would like to 
 minimize the dialplan necessary in their systems, to the point where they may 
 no longer even need the dialplan. This is a fundamentally sound idea for some 
 systems, particularly those that require scaling Asterisk out to many 
 machines.
 
 There are also some Asterisk users who build complex applications on top of 
 Asterisk, and who find having to use multiple interfaces cumbersome. They 
 like ARI, and would like to see it able to do more than what it currently 
 does today.
 

Don’t have a problem with extending the Asterisk user interface or changing 
Asterisk internals that are not visible to users. Do object to taking away 
taking away user functionality like the dial plan that users expect, are 
familiar with, and has made the Asterisk project successful.

 Fully deprecating a feature in Asterisk is non-trivial. You must have:
 (1) A logical and full replacement for the feature
 (2) Buy-off from the developer community
 (3) Several major versions of the project in which the deprecated feature 
 must remain
 
 Even in the case of point #3, deprecated features have often lasted in *many* 
 versions of Asterisk. We are enormously conservative in what we choose to 
 remove from the project.
 

Not interested in what rules or process steps need to be followed to deprecate 
features. The fact of the matter is you’re not starting with a blank sheet of 
paper and you can’t simply abandon the existing user interface because what 
will really happen is your users will abandon you and your project

Re: [asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 23, 2014, at 1:55 AM, Olle E Johansson o...@edvina.net wrote:

 It is critical that a group of developers ask themself questions along 
 these lines - what if???
 
 - What if we removed AGi and AMI?
 - What if we made a pluggable PBX?
 - What if we restarted working on a SIP channel?
 - What if we made a whole new bridge architecture?
 - What if we skip the idea of making a PBX?
 
 Good development quite frequently starts with these kind of ideas and 
 questions that may see crazy but results in really good changes.
 
 Brainstorms needs to be open and not restricted, that is what the 
 astridevcons are for. We need to go wild and see what comes out of it.
 
 A lot of the great changes we see in Asterisk 13 comes from many years 
 of wild discussions. Pinemango anyone?
 

The unacknowledged problem we’re dealing with is the fact that we’re not 
starting with a blank sheet of paper, but rather a mature user interface that 
users expect, are familiar with, and has made project successful. Extending the 
the user interface is one thing, throwing it away is another entirely different 
thing. But hey, you have the user's community attention at Astricon, why not 
have the courage of your convictions and announce to the the unwashed masses 
you’re planning to do away with the dial plan? 

 /O
 
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[asterisk-users] AstriDevCon 2014: Agenda item Deprecate AMI/AGI (Ben Klang)

[Paul Albrecht]

Really? Shouldn’t something this major affecting the entire Asterisk community 
get discussed on the lists? Any idea what Leif is talking about when he says 
the community is in transition, moving from dial plan model to external control?

Here’s a link to the notes posted on the Asterisk wiki: 
https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2014
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Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item Deprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:

 Paul Albrecht wrote:
 Really? Shouldn’t something this major affecting the entire Asterisk
 community get discussed on the lists? Any idea what Leif is talking
 about when he says the community is in transition, moving from dial
 plan model to external control.
 
 It was something Ben Klang brought up and wanted to talk about - it's 
 not something that has been decided 'nor does anyone know what the 
 future entails. Any further discussions will naturally occur on the 
 mailing list and in fact some things have explicit action items to bring 
 them up on here.
 

The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy 
talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the first 
place. It’s completely impractical and can never happen. Moreover, Leif seems 
to think we (the asterisk community) are in transition. What does that mean? 
Are we abandoning the dial plan? Seriously? That’s never gonna happen either. 
ARI isn’t easier to use than dial plan scripting. I guess one could hope that 
what happens in Vegas stays in Vegas”, but I don’t think the Asterisk 
community has that kind of luck.

 Cheers,
 
 -- 
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org
 
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Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda itemDeprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote:

 
 On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
 
  Paul Albrecht wrote:
  Really? Shouldn’t something this major affecting the entire Asterisk
  community get discussed on the lists? Any idea what Leif is talking
  about when he says the community is in transition, moving from dial
  plan model to external control.
 
  It was something Ben Klang brought up and wanted to talk about - it's
  not something that has been decided 'nor does anyone know what the
  future entails. Any further discussions will naturally occur on the
  mailing list and in fact some things have explicit action items to bring
  them up on here.
 
 
 The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy 
 talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the 
 first place. It’s completely impractical and can never happen. Moreover, Leif 
 seems to think we (the asterisk community) are in transition. What does that 
 mean? Are we abandoning the dial plan? Seriously? That’s never gonna happen 
 either. ARI isn’t easier to use than dial plan scripting. I guess one could 
 hope that what happens in Vegas stays in Vegas”, but I don’t think the 
 Asterisk community has that kind of luck.
 
 
 Just because someone decided to bring up a radical idea does not mean we 
 refuse to discuss it.
 

So you agree that deprecating AMI/AGI is “crazy talk” but you’ll discuss it 
because of your open-mindedness?

 This is an open source project. Communication is done in an open, transparent 
 manner. People should feel like they can bring up interesting, radical, and 
 yes - even crazy - ideas.
 

By the same token, when you propose ideas, you must be prepared for honest 
criticism and accept it in graciously rather than simply resorting to argument 
ad hominem. 

 If you don't like that, you don't have to participate in the discussion.
 

You haven’t really responded to the substance of my post, that is, is asterisk 
abandoning the dial plan?

 -- 
 Matthew Jordan
 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org
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Re: [asterisk-users] [asterisk-dev] AstriDevCon 2014: Agenda item DeprecateAMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 11:47 AM, BJ Weschke bwesc...@btwtech.com wrote:

 On 10/22/14, 12:14 PM, Paul Albrecht wrote:
 On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
 
 Paul Albrecht wrote:
 Really? Shouldn’t something this major affecting the entire Asterisk
 community get discussed on the lists? Any idea what Leif is talking
 about when he says the community is in transition, moving from dial
 plan model to external control.
 It was something Ben Klang brought up and wanted to talk about - it's
 not something that has been decided 'nor does anyone know what the
 future entails. Any further discussions will naturally occur on the
 mailing list and in fact some things have explicit action items to bring
 them up on here.
 
 The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy 
 talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the 
 first place. It’s completely impractical and can never happen. Moreover, 
 Leif seems to think we (the asterisk community) are in transition. What does 
 that mean? Are we abandoning the dial plan? Seriously? That’s never gonna 
 happen either. ARI isn’t easier to use than dial plan scripting. I guess one 
 could hope that what happens in Vegas stays in Vegas”, but I don’t think 
 the Asterisk community has that kind of luck.
 
 
 
  It doesn't merit discussions and shouldn't be on the agenda in the 
 first place. Really?
 
  Paul, aside from the Digium team, everyone else that's there at DevCon 
 have spent outside funds to get there and I think the consortium is 
 pretty well able to discuss whatever they'd like regardless of your 
 dictator like statements which goes against everything that an open 
 sourced project/community is supposed to be.   There have been years 
 where I'm able to attend DevCon and there have been other years, like 
 this one,  where I'm not able to attend because of prior business 
 commitments. In the prior years where I haven't been able to attend, I 
 don't personally feel like anything major was implemented without first 
 being vetted with the list and community at large. I'm not really sure 
 why you perceive the whole AstriDevCon event to be some kind of 
 conspiracy theory against the community at large, but knowing both Josh, 
 Leif, and many other people in that room for some number of years now, I 
 can assure you that I've never seen anything other than 100% transparency.
 
  You've made more than clear in prior posts to this forum that you're 
 not really a fan of ARI. I think we all get it. There are other people 
 that are fans and, for them, they're in transition to using it in a more 
 mainstream fashion because it's able to do things for them that AMI and 
 AGI cannot.  I personally still use AGI and AMI in many production 
 scenarios with Asterisk today and I'm only just thinking lately about 
 certain applications that I could transition to ARI. We cannot discount 
 that there's a very large cost for the developers, testers, and the 
 community at large to continue to keep AMI/AGI maintained and 
 functionally up to date with all the Asterisk changes along with ARI 
 given the way that AMI/AGI were originally implemented in the codebase. 
 For people that are absolutely hooked on still using AMI/AGI in the 
 longer term, perhaps a discussion could ensue at some point about a 
 bridge with AMI/AGI functionality being built off of ARI itself, or 
 maybe that's just crazy talk. I really don't know. The great thing about 
 Asterisk and the community around it is that if there's enough 
 participation and interest, anything can happen.

What you’re discounting is the Asterisk community that have used and are using 
dial plans and AGI/AMI. If ARI can’t work in that environment, then ARI should 
be abandoned as simply unworkable.

 
  BJ
 
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Re: [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 2:26 PM, Leif Madsen lmad...@thinkingphones.com wrote:

 
 
 On 22 October 2014 14:55, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote:
 This is an open source project. Communication is done in an open, 
 transparent manner. People should feel like they can bring up interesting, 
 radical, and yes - even crazy - ideas.
 
 
 By the same token, when you propose ideas, you must be prepared for honest 
 criticism and accept it in graciously rather than simply resorting to 
 argument ad hominem.  
 
 I don't think that word means what you think it means.
 
  
 
 If you don't like that, you don't have to participate in the discussion.
 
 
 You haven’t really responded to the substance of my post, that is, is 
 asterisk abandoning the dial plan?
 
 Someone proposed an idea during the devcon. Nothing has been decided or even 
 discussed yet. What you're reading and freaking out about is simply a list of 
 ideas mentioned during the devcon, and noted in a document. That's as far as 
 it has gotten; a list of minutes you're reading without further context based 
 on actual participation in the conference itself.
 

Here’s a link to the minutes:  
https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2014

It has you saying: Leif: we're in a transition, moving from dialplan model to 
external control model.  Probably need external application to be built  for us 
to move completely away from AMI/AGI.

So you’re saying Asterisk is moving away from the dial plan or were you 
misquoted?

 
 Leif.
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Re: [asterisk-users] AstriDevCon 2014: AgendaitemDeprecate AMI/AGI(Ben Klang)

[Paul Albrecht]

On Oct 22, 2014, at 2:13 PM, Scott Griepentrog sgriepent...@digium.com wrote:

  is asterisk abandoning the dial plan?
 
 It's clear that there is a desire to have a way of running Asterisk with 
 little or no dialplan.  While currently there is no way to abandon the 
 dialplan as you point out, that could actually happen, someday, many years 
 and versions from now.  But even then I would expect there could be a 
 loadable module to add dialplan support for those who still need it, where 
 the dependencies on dialplan have since been removed from the core.
 

There has to be some more justification for such a profound change to a mature 
product interface than some vague desire by unknown persons who know best for 
the entire Asterisk community. 

 So, to answer your question, yes, and no.
 
 On Wed, Oct 22, 2014 at 1:55 PM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 11:31 AM, Matthew Jordan mjor...@digium.com wrote:
 
 
 On Wed, Oct 22, 2014 at 11:14 AM, Paul Albrecht palbre...@glccom.com wrote:
 
 On Oct 22, 2014, at 10:33 AM, Joshua Colp jc...@digium.com wrote:
 
  Paul Albrecht wrote:
  Really? Shouldn’t something this major affecting the entire Asterisk
  community get discussed on the lists? Any idea what Leif is talking
  about when he says the community is in transition, moving from dial
  plan model to external control.
 
  It was something Ben Klang brought up and wanted to talk about - it's
  not something that has been decided 'nor does anyone know what the
  future entails. Any further discussions will naturally occur on the
  mailing list and in fact some things have explicit action items to bring
  them up on here.
 
 
 The suggestion that Asterisk should consider deprecating AMI/AGI is “crazy 
 talk.” It doesn’t merit discussion and shouldn’t be on the agenda in the 
 first place. It’s completely impractical and can never happen. Moreover, 
 Leif seems to think we (the asterisk community) are in transition. What does 
 that mean? Are we abandoning the dial plan? Seriously? That’s never gonna 
 happen either. ARI isn’t easier to use than dial plan scripting. I guess one 
 could hope that what happens in Vegas stays in Vegas”, but I don’t think 
 the Asterisk community has that kind of luck.
 
 
 Just because someone decided to bring up a radical idea does not mean we 
 refuse to discuss it.
 
 
 So you agree that deprecating AMI/AGI is “crazy talk” but you’ll discuss it 
 because of your open-mindedness?
 
 This is an open source project. Communication is done in an open, 
 transparent manner. People should feel like they can bring up interesting, 
 radical, and yes - even crazy - ideas.
 
 
 By the same token, when you propose ideas, you must be prepared for honest 
 criticism and accept it in graciously rather than simply resorting to 
 argument ad hominem. 
 
 If you don't like that, you don't have to participate in the discussion.
 
 
 You haven’t really responded to the substance of my post, that is, is 
 asterisk abandoning the dial plan?
 
 -- 
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 Digium, Inc. | Engineering Manager
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at: http://digium.com  http://asterisk.org
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Re: [asterisk-users] how to make voip client cannot use same username?

[Paul Belanger]

On Sun, Sep 28, 2014 at 11:51 AM, rafa alfurqan rafa.alfur...@gmail.com wrote:
 Hi All,

 I have one asterisks server and 3 client (i'm using voip sip client for my
 handphone).
 I've configured sip.conf and extension.conf with 3 user different. And
 nothing wrong with them, i could make them to make a call too.

 what i want to ask is, i was try to use 1 user (ex:1001) in 2 different
 client.
 example:
 client 1 (1001) make a call to client 2 (1002) -- ok
 then in client 3, i used (1001) same username with client 1. when client 1
 is connecting with client 2, my client 3 could make a call to with client 2
 (1002) with the same username in client 1.

 how i could make the system, so i cannot use with 1 username in 2 different
 client before i make a call (when registering process in voip client), or at
 least my voip client cannot use same username if that username is connected
 with the other user?


Since what you describe is a valid for SIP, you'll have to drop the
packets at the network level (firewall). Or use the ACL system in
asterisk to restrict it.

-- 
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Re: [asterisk-users] log caller hangup events

[Paul Greenberg]

Hi,


I am mostly concerned with inbound calls.

Would it work the same?


Regards,

Paul


From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnan N 
gopalakrishnan...@gmail.com
Sent: Monday, August 18, 2014 4:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] log caller hangup events

Hi,

You can use Hangup handler. May be this post can you help you,
http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html

Regards


On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg 
p...@greenberg.promailto:p...@greenberg.pro wrote:

All,


I would like to log a message whenever a party hangs up a call or session, i.e. 
no Dial(), user drops off a menu. The message should include the length of the 
user's session, the session's start time, and called ID.


Theoretically, I could set up a channel variable and then ...


Any advice would be most welcome!


Regards,

Paul

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Re: [asterisk-users] Error opening file for reading: Permission denied

[Paul Greenberg]

Mitch,

Is it the below error?

if ((fd = open(filename, O_RDONLY))  0) {
ast_log(LOG_WARNING, Cannot open file '%s' for reading: %s\n, 
filename, strerror(errno));
return NULL;
}

Regards,
Paul

From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Mitch Claborn 
mitch...@claborn.net
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Error opening file for reading: Permission denied

Asterisk 12.4

I am seeing message Error opening file for reading: Permission denied
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file.  Is there a way to find out which file is having
trouble?

--

Mitch


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[asterisk-users] log caller hangup events

[Paul Greenberg]

All,


I would like to log a message whenever a party hangs up a call or session, i.e. 
no Dial(), user drops off a menu. The message should include the length of the 
user's session, the session's start time, and called ID.


Theoretically, I could set up a channel variable and then ...


Any advice would be most welcome!


Regards,

Paul
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Re: [asterisk-users] Asterisk peer definition registration

[Paul Belanger]

On Sat, Aug 16, 2014 at 11:21 AM, Steve Ng steveng.1...@gmail.com wrote:
 Hi,

 I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my
 real-time, I would set the SIP credential based on what the user has
 provided.

 For example

 [name]
 type=peer
 defaultuser=USER_PROVIDED
 secret=USER_PROVIDED
 host=USER_PROVIDED

 When I reset Asterisk, Asterisk will attempt to register with the sip
 provider. And if there are sufficiently amount of records with invalid
 credentials, I'll get blocked by the SIP provider as they might think that
 I'm brute forcing.

 Just a question to check if there's any chance I could ask Asterisk not to
 register when I reset. Or is there any other possible solution for this?

No, only reload after your ITSP brute force timer has expired.

-- 
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Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Paul Belanger]

On Fri, Aug 15, 2014 at 10:41 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
 Hello,

 After having thought this through a bit I have some thoughts I'd like to
 share.

 In this case where the rtp profile is RTP/AVP Asterisk accepts and handles
 the call normally. If a webrtc client calls a sip client, or even another
 webrtc client, rtpengine is needed to step in (in my setup most of the
 clients would indeed be webrtc, but some of them might be sip). I think it
 would be better to use RTP/SAVPF throughout the process if both clients are
 webrtc (or otherwise speak RTP/SAVPF), but currently there is no way to
 accomplish this?

 Is it possible to configure Asterisk to only accept the RTP/SAVPF profile,
 and send 488 to all others? If it's not possible to force Asterisk to ignore
 rtp profiles (thus allowing the sdp be handled by rtpengine entirely), I'd
 prefer to use RTP/SAVPF or RTP/SAVP in the communication between Kamailio
 and Asterisk servers and use rtpengine to bridge to RTP/AVP and RTP/AVPF
 only if the client cannot speak securely.

 I'd very much like to hear opinions and thoughts on these.

Again, I'll only share my experiences, but we do the complete
opposite.  Traffic between kamailio and asterisk is only RTP/AVP since
the version of asterisk we are using does not support RTP/SAVPF (1.8).
However, if you want RTP/SAVPF then honestly, you should just
completely remove rtpengine from the picture since newer version of
asterisk support both RTP/AVP and RTP/SAVPF (asterisk 12+).

What I think you should do is go back to the basics, and document
everything you want to do.  Right now you have too many pieces in the
puzzle and making the setup complicated.  Like I said before, this is
a complex setup and you need to start some place.  Here is a diagram
of what we do.

webrtc (RTP/SAVPF) - kamailio - rtpengine  - asterisk (RTP/AVP)

This way, only RTP/AVP is in the core of our network. Rtpengine is on
the edge (where it belongs), proxing rtp traffic.  And, for us, we
keep RTP/SAVPF outside of asterisk since support for it has been
recently added. I also believe there are some open issue with dtls +
srtp too.

-- 
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Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Paul Belanger]

On Fri, Aug 15, 2014 at 12:17 PM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
 Thanks Paul, I appreciate your thoughts.

 I understand your way, it's logical in your environment. I prefer to use LTS
 versions of Asterisk so I'm guessing what I want to do is not quite possible
 with Asterisk 11.

 I'd prefer my setup to work like this in different cases.

 webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- webrtc (rtp/savpf)
 sip (rtp/avp) -- kamailio -- rtpengine (rtp/savpf) -- asterisk -- kamailio
 -- rtpengine (rtp/avp) -- sip (rtp/avp)
 webrtc (rtp/savpf) -- kamailio -- asterisk -- kamailio -- rtpengine
 (rtp/avp) -- sip (rtp/avp)

 ... essentially, using RTP/AVP only when the client does not speak securely.

 It appears I'll have to try out the RTP/AVP way until there is an Asterisk
 that can accomplish this without having to use peer-specific settings.
 Down-side to this is that rtpengine needs resources from the server for
 webrtc clients even though both ends speak the same profile.

 It's not so complicated now that I know more on what Asterisk supports and
 how it handles the sdp, I just needed to learn by doing, testing and asking.
 I must be a bit ahead of my time for going for a RTP/SAVPF within my
 architecture, but using RTP/AVP is not such a bad option as srtp is on its
 way anyway in future Asterisk versions and the rtp flowing between Kamailio
 and users' networks are far more important than internal rtp traffic.

Fair enough, I won't be able to help moving forward.  We opted for
only using RTP/AVF with asterisk because how new the code for RTP/AVPF
and dtls-strp handling is.  And since RTP/AVF has been around since
the start, it is pretty stable. And this is the primary reason people
are using rtpengine with asterisk to start.  So, in your setup listed
above, rtpengine is not needed, since newer versions of asterisk
support both. Adding it in will just complicate your setup.

-- 
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [asterisk-users] Asterisk on CentOS7

[Paul Greenberg]

building from source.


From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Anthony Messina 
amess...@messinet.com
Sent: Thursday, August 14, 2014 12:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk on CentOS7

On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote:
 Hi Anthony,

 That script does not work. My guess is that it is related to the way
 asterisk interacts with CentOS environment.

 Best Regards,
 Paul Greenberg, Esq.

 On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote:
   I installed CentOS 7 on a spare server along with all our Asterisk
 
  configuration system and the only thing that failed is the asterisk
  startup script included in the asterisk tarball.  I guess because the
  startup system has changed so much that script will have to be updated.
  Everything else worked fine as far as I can tell but obviously I did not
  stress test that installation.

 You can use the systemd unit file I have here:
 https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633
 881bb4d1baaeb6ae1e63c032abdc

 It's what Fedora uses as well.  This should work properly in EL7.  Hopefully
 in not too long, I'll have Asterisk 13 builds for EL7, though I need to
 figure out a few dependency issues: https://messinet.com/rpms/

I do know that the Fedora EPEL project provides Asterisk for EL6, and I
believe they will support it for EL7 as well, once EPEL 7 comes out of beta
status.  EL7 uses systemd, so I'm not sure that the regular init file will
work properly without some tweaking, which is why I pointed you to the systemd
unit file that is used by the Asterisk RPMs from Fedora 20, and the one I use
with the RPM builds I make myself.

How are you installing Asterisk on CentOS 7?  Are you doing a regular
make/install from source, or using RPM packages?

--
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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [asterisk-users] Asterisk on CentOS7

[Paul Greenberg]

Hi Matthew,

I am using it. Works like a charm!

Running it for 3 week already and have no issues. However, my system is not 
heavily utilized, i.e. 50-150 phone calls a day.

The only thing is I was not able to get asterisk integrated with CentOS 
services daemon. So, I am starting asterisk manually.

Best Regards,
Paul Greenberg, Esq.

Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
E-mail: p...@greenberg.pro
Tel:  201-402-6777
Fax:  201-301-8876
Web: http://www.greenberg.pro



From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Matthew Jordan 
mjor...@digium.com
Sent: Wednesday, August 13, 2014 12:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk on CentOS7

On Wed, Aug 13, 2014 at 3:10 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi

 Is anyone using asterisk on CentOS 7?

 If so, is it working fine and as expected?


Random data point: the Asterisk project's build agents are still on CentOS 6.

Your mileage may vary.

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

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Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Paul Belanger]

On Wed, Aug 13, 2014 at 4:35 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
 Hi,

 Wow, thanks Paul, realizing the problem makes a lot of sense.

 So I setup Kamailio as a peer, but if I disable chan_sip module completely,
 I can't do it in sip.conf like I'd otherwise assume to do. I tried to
 rebuild Asterisk without chan_sip, but I guess that's not quite the way to
 go? Asterisk stopped sending back any sip messages so either there is a
 configuration means on how to do this or I'm doing something wrong with my
 current setup. My next thought was to compile Asterisk normally and set
 rtcachefriends to no, that did not work either, when dialing the cli stated:
 app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP'
 (cause 20 - Subscriber absent)
 which I guess says Asterisk does not know where to send the message.

 The inner workings of Asterisk is a bit beyond me, if you don't mind giving
 advice on how to proceed I'd be most grateful.

I think you are still mis-understanding me.  I'll try to be clearer.

From the POV of asterisk, you do still need chan_sip, however the only
peer asterisk needs to be away of it Kamailio.  All other peers will
be stored within kamailio.  This was the reason for my comment about
realtime sip, you don't need it.

Then, within kamailio, you'll need to invoke rtpengine using
(rtpproxy-ng with kamailio 4.1) to rewrite the sdp for the invite to
asterisk.  You'll use the rtpproxy_offer and rtp_answer functions to
remove ICE when calls originate from webrtc clients.  Since you are
not using a websocket in asterisk, it will just be a SIP over udp, the
need for ICE and SAVPF is not needed.

What you are trying to do is pretty complicated, it took me about 2
weeks to get everything setup properly.  There is good information[1]
on the web, you just need to google for it.

[1] http://www.slideshare.net/crocodilertc/webrtc-websockets

-- 
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Re: [asterisk-users] Asterisk on CentOS7

[Paul Greenberg]

Hi Anthony,

That script does not work. My guess is that it is related to the way asterisk 
interacts with CentOS environment.

Best Regards,
Paul Greenberg, Esq.

Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
Tel:  201-402-6777
Fax:  201-301-8876
Web: http://www.greenberg.pro


From: asterisk-users-boun...@lists.digium.com 
asterisk-users-boun...@lists.digium.com on behalf of Anthony Messina 
amess...@messinet.com
Sent: Wednesday, August 13, 2014 1:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk on CentOS7

On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote:
  I installed CentOS 7 on a spare server along with all our Asterisk
 configuration system and the only thing that failed is the asterisk
 startup script included in the asterisk tarball.  I guess because the
 startup system has changed so much that script will have to be updated.
 Everything else worked fine as far as I can tell but obviously I did not
 stress test that installation.

You can use the systemd unit file I have here:
https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633881bb4d1baaeb6ae1e63c032abdc

It's what Fedora uses as well.  This should work properly in EL7.  Hopefully
in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure
out a few dependency issues: https://messinet.com/rpms/

-A

--
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8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Paul Belanger]

On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
 Hello,

 Thank You Paul for your reply,

 The registrations in my setup are not duplicated, the 'secret' field in the
 realtime table is empty, which causes Asterisk to not authenticate requests
 from my Kamailio. Kamailio handles registrations, and also routes the
 traffic to Asterisk using dispatcher. Also, all peers have the Kamailio
 ip:port as outbound proxy so all traffic goes through Kamailio.

That is your issue, stop using chan_sip with realtime (using data from
kamailio).  The only SIP peer asterisk should know of is kamailio, and
your webrtc clients should be anonymous SIP users.  This way, Asterisk
doesn't even need to deal with websockets and RTP/SAVPF (this is what
kamailio and rtpengine) is for.

In your current setup, you are bypassing the functionality of
rtpengine and not even leveraging it.

 Looks like version 11.11 works differently, I'll try to revert back to a
 previous version, and see if that works. I know at least the 'force_avp'
 field is new to 11.11 so it's safe to assume there's some difference between
 versions in rtp profile handling.

 It would be good to know how to handle this scenario in the new versions as
 well, I'll probably need to upgrade ahead anyway.



-- 
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Re: [asterisk-users] Asterisk 12 on Debian Wheezy

[Paul Belanger]

On Tue, Aug 12, 2014 at 12:38 PM, Olivier oza.4...@gmail.com wrote:
 Hello,

 A couple of questions in relation with Asterisk 12 on Debian Wheezy.

 1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
 the sole binary to add PJSIP stack to Asterisk 12 (compiled from
 source) ?

Yes, you should be able too.

 2. When compiling PJPROJECT from source (see
 https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject?src=search)),
 where should PJSIP .so files be located in an appropriately managed
 Wheezy system  ?
 In other words should I get the line bellow or something else ?
 libpjsua.so (libc6) = /usr/lib/libpjsua.so

You will likely need to pass the pjproject directory to configure.

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Re: [asterisk-users] Letting rtp profiles be handled by rtpengine instead of Asterisk

[Paul Belanger]

On Mon, Aug 11, 2014 at 4:45 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:

 Hello,

 I'm trying to get calls working between websocket clients and sip clients.
 For clients I have sip.js based clients on chrome, Zoipers and a Grandstream
 phone. Challenge here is I'd like to have Kamailio and rtpengine to handle
 the bridging between different rtp profiles but Asterisk changes them in the
 sdp bodies along the way. I'm using Asterisk 11.11.0.

 Is there a way to configure Asterisk to ignore the rtp profile but allow
 calls to pass with either of those profiles (even though clients might
 answer with 488 which would be caught and handled by Kamailio and
 rtpengine)? In my setup I have Asterisk Kamailio realtime integration, and
 the second goal is to be able to add peers to the db table with similar
 data, as in no different values based on what kind of client wants to
 register. I'd like to allow the user to register using which ever client
 they choose (in this case one of the 3 I mentioned).

 Previously I had problems like 'rejecting secure audio stream without
 encryption details', no audio or BYE messages sent immediately after call
 has begun etc, but according to sip.js documentation
 (http://sipjs.com/guides/server-configuration/asterisk/) the settings avpf
 and force_avp affect the way Asterisk handles the rtp profiles and now my
 calls do work ok but I'd need to move the rtp profile handling to rtpengine.

We are successfully using kamailio / rtpengine with websockets and
asterisk 1.8. First question is why are you duplicating registrations
within asterisk?  Secondly, why are you using websockets in asterisk?

Without knowing more about your use case, I'll tell you how we did it.
Like I said, kamailio is responsible for our SIP/ws subscribers and
registrations.  Once within kamailio we simply dispatch traffic to
asterisk via SIP/udp.  RTP is handled by rtpengine (using rtproxy-ng)
and that is basically it.

No special configuration is needed for asterisk (in fact 1.8 has no
support for RTP/SAVPF) so we rewrite SDP on 488.  Then setup a
kamailio peer and away you go.

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[asterisk-users] Asterisk IP7960 and MWI Issue

[Paul Greenberg]

All,


I am running the following setup:

Linux 3.10.0-123.4.2.el7.x86_64 #1 SMP Mon Jun 30 16:09:14 UTC 2014 x86_64 
x86_64 x86_64 GNU/Linux
Asterisk 12.4.0

Cisco IP Phone 7960G


I have an issue with MWI. For some reason after I delete my voicemail messages, 
the MWI of the phone is ON for another 20-30 minutes. It seems that there is a 
poll interval of some kind. I am not sure whether it is a setting on the phone 
or the asterisk.


Any  ideas?


Best Regards,
Paul Greenberg, Esq.

Law Office of Paul Greenberg
530 Main Street, Suite 102
Fort Lee, NJ 07024
E-mail: p...@greenberg.pro
Tel:  201-402-6777
Fax:  201-301-8876
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Re: [asterisk-users] Notification when queue member's phone rings

[Paul Belanger]

On Wed, Jul 2, 2014 at 4:11 PM, Mitch Claborn mitch...@claborn.net wrote:
 Short question: how to get control or notification (gosub, macro, AGI) when
 a queue member's phone starts ringing due to an incoming call from the
 queue.

 Backround:  Our phone operators serve both an asterisk call queue and a
 queue for web chat support.  I have a gosub on the queue that calls to our
 app server to mark the operator unavailable for web chat as soon as they
 answer an incoming queue call.  Similarly, when a web chat is connected, it
 uses AMI to tell asterisk to take the operator out of the phone queue.

 The other day, one operator got a web chat that came in while her phone was
 ringing with a queue call, so that neither remove from queue operation was
 effective in time.  If I could get notification when the phone starts
 ringing I can reduce the window of opportunity for that by several seconds.

 It's only happened once in 2 years that I know of, so may not be worth
 worrying about.

AMI will raise the AgentCalled[1] event.

[1] 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_AgentCalled

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Re: [asterisk-users] Live Recording on the Storage Server?

[Paul Belanger]

On Thu, Apr 17, 2014 at 7:25 AM, binary dreamer
dreamer.bin...@gmail.com wrote:
 hi. I would not do that due to network issues.
 My approach is to record everything locally and every hour or so to move
 everything to a storage.

+1 save yourself the headache and do this.

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Re: [asterisk-users] Asterisk 11 under VMware?

[Paul Belanger]

On Fri, Apr 4, 2014 at 1:30 PM, Carlos Chavez cur...@telecomabmex.com wrote:
 I have found Asterisk using only SIP is very responsive on virtual
 machines.  We have used VMs for call center applications and for complex IVR
 solutions without problems.  Obviously there is overhead running a VM so you
 can never expect a VM to perform as well as bare metal.  Running a single VM
 on a server is a complete waste of resources, might as well run natively.

Well, regardless of how many VMs you run on bare metal, you do get the
benefit of the VM technology.  Even if OP runs 1 VM on the box, he
could leverage snapshots in VM ware for the purpose or migrating or
back ups.  I don't think it is a waste per say, just different
requirements.

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Re: [asterisk-users] Asterisk CLI Banner

[Paul Belanger]

On Fri, Mar 28, 2014 at 2:39 PM, Steve Edwards
asterisk@sedwards.com wrote:
 On Fri, 28 Mar 2014, Richard Kenner wrote:

 And this certainly may vary from jurisdiction to jurisdiction.  For a
 (quite dated at this point) discussion of this issue from a US perspective,
 see


 http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157


 The publication (43 pages) is dated 1988. The DMCA (1998) and subsequent
 legislation may have changed the landscape.

 My (ignorant) opinion -- just don't. Is it worth the effort to research? Is
 it worth paying a lawyer to research it and give an opinion that may be
 worth nothing until it is examined in court?

 If you want to display something custom, how about a 'wrapper' script that
 displays a file using 'curl' before handing off to Asterisk -- easier to
 implement, easier to maintain, no legal BS to consider.

 Or can you express your creativity by fiddling with ASTERISK_PROMPT?

If you really want to do it:

1) create a wrapper to asterisk -r
2) pipe the welcome message to /dev/null
3) ???
4) profit

you didn't modify Asterisk.

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Re: [asterisk-users] asteriskdocs.org says 3rd ed. is latest

[Paul Belanger]

On Mon, Mar 24, 2014 at 8:54 AM, Matt Behrens m...@zigg.com wrote:
 I made myself look a little silly recently in a talk regarding 
 asteriskdocs.org.  I didn't realize the 4th ed. of the Definitive Guide was 
 apparently actually out (http://shop.oreilly.com/product/0636920025894.do), 
 because I went by asteriskdocs.org's claim that it was being worked on in 
 OFPS (now retired, apparently.)

 Is the 4th ed. available to read online like the 3rd ed. was?  Is someone on 
 this list able to update asteriskdocs.org with current info regardless?

I pinged Leif Madsen on this and he's updating the site now. Currently
only the 3rd edition is published online.

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Re: [asterisk-users] AMI Proxy

[Paul Belanger]

On Mon, Mar 24, 2014 at 6:17 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
 Hi people

 Just having a quick check to see if anyone is using any AMI proxies and
 which are the most popular. For our purposes it must be able to connect to
 multiple asterisk instances.

All depends on the language you want to use. We used starpy for a
while, but ended up rewriting our own version.  Currently we're
connecting AMI to a message bus and passing events across the bus.

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Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

[Paul Belanger]

On Fri, Mar 21, 2014 at 11:58 PM, Al lists asteris...@gmail.com wrote:
 looking more into this, looks like this is not a issue, its related to users
 changing voicemail password from handset, asterisk rewrites the file.

Right, use passwordlocation = spool, create a secret.conf for each
mailbox, now when a user changes their password, secret.conf gets
updated not voicemail.conf.

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Re: [asterisk-users] AppVoicemail overwrites voicemail.conf

[Paul Belanger]

On Fri, Mar 21, 2014 at 3:22 PM, Al lists asteris...@gmail.com wrote:

 We noticed issues with voicemail and somehow looks like voicemail.conf has
 been overwritten:

 ;!
 ;! Automatically generated configuration file
 ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
 ;! Generator: AppVoicemail
 ;! Creation Date: Thu Mar 20 06:48:16 2014
 ;!


 i saw a bug for 1.4 and realtime but our version is 10.12.3 and we are not
 using realtime.
 anyway to prevent AppVoicemail ro auto generate files?

passwordlocation = spooldir

Read voicemail.conf about how to use it.

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Re: [asterisk-users] Need more meetme users -- hitting some limit

[Paul Belanger]

On Fri, Mar 21, 2014 at 11:53 AM, Steve Edwards
asterisk@sedwards.com wrote:
 I'm trying to determine the capacity of my host running Asterisk 11.8.1 on
 CentOS 6.5.

 The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet.

 The primary application will be bridging groups of users using meetme().

 I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1
 behaving a bit more like a production box -- bridging calls (box2).

 The call file on box1 originates a call to box2 and then plays a 2 hour WAV
 file.

 The dialplan on box2 drops the call into a meetme, creating the room name
 from the last 2 digits of the current call count -- distributing the calls
 into 100 meetmes.

 When I run a script to create 500 call files on box1, box2 starts
 complaining at 312 calls, logging 'Unable to open DAHDI pseudo channel:
 Cannot allocate memory' on the console.

 From the 'callers perspective' the call is dropped between 'There are
 currently x other participants in the conference' and the 'beep-beep.'

 'top' says Asterisk is only using about 1/2 gigabyte of RAM.

 'top' says Asterisk is using about 250% of the CPU (4 physical, 8 logical
 cores).

 'ulimit' (added to /usr/sbin/safe_asterisk in the run_asterisk() function)
 says the open file limit is 397,006.

 'ls -l /proc/$(cat /var/run/asterisk/asterisk.pid)/fd | wc -l' says Asterisk
 only has 2,194 files open.

 'iftop' sees about 24Mb of bandwidth in each direction between the boxes.

 Using confbridge() I can easily get 3,000 calls (14,869 open files, 180Mb
 bandwidth), but I'd lose some functionality and have to re-write parts of my
 application.

 Any clues of what limit I'm hitting and how to increase it?

DAHDI has a pseudo channel limit of 512, somebody has already posted
how to change it with modprode.

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

[Paul Belanger]

On Tue, Mar 18, 2014 at 1:02 PM, James Sharp ja...@fivecats.org wrote:
 Putting a whole bunch of people into a listen-only/muted Confbridge
 conference or getting the broadcaster audio into a MOH class and then just
 having callers attach to that MOH class?

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

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

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

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Re: [asterisk-users] High Availability with Asterisk

[Paul Belanger]

On Thu, Mar 6, 2014 at 10:57 AM, Mitul Limbani mi...@enterux.in wrote:
 Hello,

 Using Single Server with multiple VMs essentially kills the purpose, coz it
 doesnt protect against physical hardware failures.

 To save costs, use low end box as failover, to keep u in business, till
 primary box goes live.

Correct, in this case para-virt is not the way to go. You'll want to
use a virtualization platform that does support multi-hardware with
live migration support.

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Re: [asterisk-users] High Availability with Asterisk

[Paul Belanger]

On Thu, Mar 6, 2014 at 3:33 PM, Markus unive...@truemetal.org wrote:
 Hi Thorolf,

 Am 06.03.2014 16:21, schrieb Thorolf Godawa:

 Using (para-)virtualization with Xen could be an other option, on
 systems with low load this works reliable, but what happens on systems
 with high load? Are there any issues known about problems with the
 realtime, packet loss etc. because it runs in a VM?


 hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this
 related to high availability? I think it's not. :)

 I think the way to go for high availability (and scalability) is Kamailio!
 In a redundant setup, running on 2 separate physical machines (maybe in a
 VM, doesn't matter). Then you make them failsafe using whatever tool(s)
 available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio
 and any of them could fail (but 1 :-) ) and you will still be online.

 If you want to further develop the high availability thought, then you could
 use CephFS which will give you self-healing, 100% available storage over
 multiple physical storage servers. There you could store your Asterisk
 config files, or your MySQL database used by all the Asterisk servers, for
 CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for
 Asterisk / MySQL. :)

 And, to scale and to make the Asterisk nodes redundant (redundancy is not
 really needed anymore, since Kamailio takes care of that, but basically then
 you get also VM/physical redundancy), you could look into OpenNebula which
 provides a nice auto-scaling feature already out of the box. If there's load
 on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs
 (probably on different physical servers, otherwise it doesn't make that much
 sense performance-wise) which will automagically receive requests/calls from
 Kamailio. If the load goes down, the VM can be automagically stopped again
 to free resources for other VMs/applications. OpenNebula is less popular
 than OpenStack, which seems to be the first choice for Cloud-stuff today,
 but what I liked about OpenNebula is that it provides the auto-scaling
 feature already in the customer-facing web-frontend out-of-the-box, unlike
 OpenStack. So you could offer your customers a self-managed, redundant
 Asterisk cloud or something like that. :)

 In theory, this combination should give you a 100% redundant, auto-healing,
 auto-scaling VoIP setup. :)

+1 to this post.  A lot of good information here.

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Re: [asterisk-users] Updating to 11.7.0

[Paul Hayes]

On 19/12/13 17:15, David Lee (digium) wrote:


On Dec 19, 2013, at 10:34 AM, Jerry Geis ge...@pagestation.com
mailto:ge...@pagestation.com wrote:


[snip]


Looking that up, it says add to asterisk.conf
[options]
live_dangerously = yes

After doing this, and stopping and starting I
still get the message.



I'm having the same issue, even with it set to yes and restarting 
Asterisk, it still give that warning.



Whats up?


You want to avoid danger, so set live_dangerously = no.



I appreciate that and I do understand why but that setting doesn't work 
as described, it seems to do nothing.


While we're at it, what's the recommended alternative method to replace 
using asterisk -rx in bash scripts now?


cheers,
Paul.


Jerry


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Re: [asterisk-users] Updating to 11.7.0

[Paul Hayes]

On 05/03/14 12:56, Paul Hayes wrote:




I appreciate that and I do understand why but that setting doesn't work
as described, it seems to do nothing.

While we're at it, what's the recommended alternative method to replace
using asterisk -rx in bash scripts now?

cheers,
Paul.



Apologies for replying to my own post but I since found this:

https://issues.asterisk.org/jira/browse/ASTERISK-23084

and after testing it turns out that even though asterisk -rx always 
shows the warning, the command is still executed so no where near as bad 
a problem as I initially thought.


I guess the live_dangerously setting wont eventually have an affect on 
asterisk -rx (it doesn't really make any sense to since if someone is 
already in as root then you are screwed!).


cheers,
Paul.

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Paul Belanger]

On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote:

 On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 
 No such thing as 'free open source g729 license', if you actually read the
 site:


 There is regarding the copyright on the code. The fact it is also patent
 encumbered is a different issue.

 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
 patent holders for using their algorithm.

 So, basically you are illegal using them if you didn't pay for them.


 Not true. He said it was a lab setup. It is totally legit to use patented
 processes in an evaluation/lab environment.

Correct, I didn't mention this, since I was assuming OP was talking
about getting it into production.  Should have been more clear.

  3) Is there a performance/stability/security gain when using the
  commercial vs. open source version or vice versa.
 
 See above about about open source license.

 Your comment about open source is irrelevant to performance, stability, and
 security. WRT these criteria, I would be surprised if there is much of a
 difference. The free software isn't locked to a mother board, so that might
 count towards performance by some measures.

 Now having said that. I agree once you leave the lab environment and decide
 you need g.729, you will unfortunatly need a licence to keep using it.

 The real question is: is there really any choice other than Digium for the
 licence? Due to the dual licensing of the asterisk code, even if you could
 license the codec elsewhere, you might be violating Digium's OSS license
 when you don't but their commercial asterisk license.

 Cheers,
 Darryl


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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Paul Belanger]

On Thu, Feb 27, 2014 at 5:24 PM, Jayson Devor jayson.de...@gmail.com wrote:
 Hello Everyone,

 We are looking to transition our 23 channels from testing/lab into
 production. During testing we used the free open source g729 license
 using the instructions found here:

 http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/

No such thing as 'free open source g729 license', if you actually read the site:

DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
patent holders for using their algorithm.

So, basically you are illegal using them if you didn't pay for them.

 A little more about our setup. All recordings have been converted to
 G729, all voicemail messages are also in G729, finally allow=g729,
 disallow=all is specified in our config.

 Questions:

 1) Is there anything we overlooked in our attempt to implement g729
 passthough, and stop all transcoding efforts?
 2) do we still need to purchase 23 G729 licenses? If so, is asterisk
 10$ license recognized by the patent holders (ie, is Digium authorized
 to sell the license on behalf of the patent holders)?
Yes, getting a license from digium should be sufficient to cover your
usage.  Plus you'll be supporting the project.

 3) Is there a performance/stability/security gain when using the
 commercial vs. open source version or vice versa.

See above about about open source license.

 I was reluctant to bring this topic up yet again , and yes I did
 google around and read the different material on the subject however,
 I am still in need of some definitive answers.


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Re: [asterisk-users] SIP 603 Declined error message

[Paul Belanger]

?Set(CALLERID(name)=)) in new stack

 -- Executing [cidlookup_return@cidlookup:2]
 Return(SIP/trunk503in-010b, ) in new stack

 -- Executing [51104@from-trunk:4] ExecIf(SIP/trunk503in-010b, 0
 ?Set(CALLERID(name)=3145152244)) in new stack

 -- Executing [51104@from-trunk:5] Set(SIP/trunk503in-010b,
 __CALLINGPRES_SV=allowed_not_screened) in new stack

-- Executing [51104@from-trunk:6] Set(SIP/trunk503in-010b,
 CALLERPRES()=allowed_not_screened) in new stack

 -- Executing [51104@from-trunk:7] Goto(SIP/trunk503in-010b,
 app-blackhole,hangup,1) in new stack

 -- Goto (app-blackhole,hangup,1)

 -- Executing [hangup@app-blackhole:1] NoOp(SIP/trunk503in-010b,
 Blackhole Dest: Hangup) in new stack

 -- Executing [hangup@app-blackhole:2] Hangup(SIP/trunk503in-010b,
 ) in new stack

   == Spawn extension (app-blackhole, hangup, 2) exited non-zero on
 'SIP/trunk503in-010b'

 Scheduling destruction of SIP dialog '8066eb6f589ce3125b652973b4b00' in
 32000 ms (Method: INVITE)



 --- Reliably Transmitting (NAT) to 172.17.184.46:31285 ---

 SIP/2.0 603 Declined

 Via: SIP/2.0/TCP
 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285

 Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00

 From: Haley, Scott
 sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00

 To: sip:51...@edj.devjones.com;tag=as06e2e068

 Call-ID: 8066eb6f589ce3125b652973b4b00

 CSeq: 1 INVITE

 Server: FPBX-2.8.1(1.8.13.0)

 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
 PUBLISH

 Supported: replaces, timer

 Content-Length: 0





 



 --- SIP read from TCP:172.17.184.46:31285 ---

 ACK sip:51...@edj.devjones.com SIP/2.0

 From: Haley, Scott
 sip:3145152...@edwardjones.com;tag=8066eb6f589ce3124b652973b4b00

 To: sip:51...@edj.devjones.com;tag=as06e2e068

 Call-ID: 8066eb6f589ce3125b652973b4b00

 CSeq: 1 ACK

 Max-Forwards: 70

 Via: SIP/2.0/TCP
 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285

 User-Agent: Avaya CM/R016x.02.0.823.0

 Route: sip:192.168.122.51;transport=tcp;lr;phase=terminating

 Content-Length: 0



 -

 --- (10 headers 0 lines) ---


You'll want to talk to the FreePBX guys, as you are just hanging up
the outbound call.

-- Executing [51104@from-trunk:7] Goto(SIP/trunk503in-010b,
app-blackhole,hangup,1) in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1]
NoOp(SIP/trunk503in-010b, Blackhole Dest: Hangup) in new stack
-- Executing [hangup@app-blackhole:2]
Hangup(SIP/trunk503in-010b, ) in new stack

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Re: [asterisk-users] G729 - what happens if licences used up?

[Paul Belanger]

On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote:
 I haven't been able to find the answer online, and am not currently
 able to conduct an experiment to find the answer...

 I understand that in a SIP call where G729 has been negotiated as the
 preferred codec, a G.729 licence is not consumed until there is a need
 to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
 or enter a Meetme, etc.

 What happens when a SIP call in progress needs a G.729 licence and
 they are all in use already? Does the call fail, or go silent, or do a
 re-INVITE to negotiate another codec?

 I'm interested in what happens on Asterisk 1.2 (for a legacy system),
 and also whether it is any different on later versions.

The question depends if you are offering up other codecs or not.  If
you only using g729, the call will fail to establish because lack of
codecs.  If you offer a both g729 and ulaw, then ulaw will be used.

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Re: [asterisk-users] Changing gateway address

[Paul Belanger]

On Fri, Feb 14, 2014 at 5:40 PM, Dave Swangler ctit...@live.com wrote:
 Hello,

 I inherited an Asterix phone system. I am well versed in Windows based
 platforms but have zero experience in Linux and Asterix, no make matters
 worse I have no documentation on this system. I had to change the entire
 networks gateway address for various reasons but now the Asterix system will
 not send messages via email. I think it is because of the gateway change.
 How do I change the gateway address? Is this product something I could
 contract out to have remote support?  Thanks,

What you describe is more of a Linux support issue then specific to
Asterisk.  Depending on your OS, will dictate how to change your
gateway.

check /etc/network/inferfaces if you are ubuntu / debian.

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Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

[Paul Belanger]

On Thu, Feb 13, 2014 at 1:04 AM, George Joseph
george.jos...@fairview5.com wrote:
 On Wed, Feb 12, 2014 at 6:26 PM, Paul Belanger
 paul.belan...@polybeacon.com wrote:

 On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote:
  Hello,
 
  How does extensions.lua compares to extensions.conf or extensions.ael on
  stability, performance and features ?
 
  Would you recommand  extensions.lua as an easy/easier way to access
  memcached, redis or equivalent ?
 
  Thoughs ? Comments ?
 
 The lack of replies should give you your answer.  Extensions AEL and
 LUA don't get much action these days, I'm sure there are a few people
 that use them but extensions.conf has way more code coverage from a
 testing POV.

 Your better off using AGI if you want to leverage redis or memcached.

 Actually, I use Lua dialplans in several production systems.  Some are used
 in conjunction with traditional dialplans and some are the only source of
 dialplans.  They've always been rock solid.   I actually find it easier to
 configure even a moderately complex dialplan than the traditional dialplan
 syntax.

Cool, you are in the minor on that one.  My only caution about using
them about be the lack of support if you had issues.

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Re: [asterisk-users] How does extensions.lua compares to extensions.conf ?

[Paul Belanger]

On Wed, Feb 12, 2014 at 12:50 PM, Olivier oza.4...@gmail.com wrote:
 Hello,

 How does extensions.lua compares to extensions.conf or extensions.ael on
 stability, performance and features ?

 Would you recommand  extensions.lua as an easy/easier way to access
 memcached, redis or equivalent ?

 Thoughs ? Comments ?

The lack of replies should give you your answer.  Extensions AEL and
LUA don't get much action these days, I'm sure there are a few people
that use them but extensions.conf has way more code coverage from a
testing POV.

Your better off using AGI if you want to leverage redis or memcached.


-- 
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[asterisk-users] answering machine screening with MixMonitor

[G. Paul Ziemba]

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

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

I am currently using MixMonitor to append the audio to a
named pipe (mkfifo /home/asterisk/var/soundpipe.au), as
follows (extensions.conf):

[from-pstn]
exten = s,1,Wait(20)
exten = s,2,Answer
exten = s,3,Wait(1)
exten = s,4,Set(TIMEOUT(response)=10)
exten = s,5,MixMonitor(/home/asterisk/var/soundpipe.au,av(2))
exten = s,6,Background(/home/asterisk/sounds/main)

I wrote a small script to loop opening the named pipe and then
copying to /usr/bin/play from the sox package.

MixMonitor uses the filename extension to determine what format
to use for the audio data. I tried all of the formats common
to both asterisk (core show file formats) and play (as described
on the man page), except for raw. I'm not sure if raw is
compatible with sox play - I need to determine bit rates and
other parameters before I can try it.

I found that in all cases I tested, there was significant latency
between the audio on the PSTN line vs. the audio played from the
speakers, on the order of 3-10 seconds depending on the format
specified. Based on some debug output from my pipe-reader script,
it seemed that asterisk opened the write end of the pipe immediately
but did not start writing data until the aforementioned delay had
passed. au and sln had the lowest latency (3 seconds), so I'm
using au for now. 

Is there any way to reduce the startup latency and make MixMonitor
write the audio stream to the output file immediately? I looked
briefly at apps/app_mixmonitor.c and main/file.c but I don't fully
understand the code. Is mixmonitor forking an external conversion
process to generate the audio data?

thanks for any insights!
-- 
G. Paul Ziemba
FreeBSD unix:
 9:06AM  up 10 days, 11:05, 4 users, load averages: 1.39, 1.50, 1.54

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Re: [asterisk-users] Asterisk as a media gateway

[Paul Belanger]

On Fri, Jan 31, 2014 at 11:27 AM,  richard.seg...@marisec.ca wrote:
 I'm playing around in a lab, and I was wondering if its possible to have 
 Asterisk act similar to that of a Avaya PBX, where we have media gateways do 
 the heavy lifting.

 This is what I was thinking of trying.

 1.  One asterisk server will contain the logic of the phone system (ex: 
 queues, extensions...etc).

 2.  The mains server will not handle RTP traffic,  it will send the RTP 
 traffic to another system (another asterisk box?) for processing.

 At the end of the day, what I am hoping for is to have 1 brain, and mutiple 
 work horse audio gateways that can be added and removed as needed.

 Has this been done?  Can anyone point me to some documentation on how others 
 have done this?

 It's always fun to play

Yes, this is basically functionality for Asterisk. If you are using
SIP, you want to REINVITE media away from your core Asterisk box.  I
suggest picking up the book[1] and reading the chapter on connecting
multiple Asterisk boxes together.

[1] http://www.asteriskdocs.org/

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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Paul Belanger]

On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
dcunning...@voisonics.com wrote:
 Hi Paul,

 Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
 and arriving at the Asterisk server. This is why it's a mystery that
 Asterisk doesn't see the call coming in. We tried removing the firewall (so
 iptables -L shows no rules at all) but that didn't help unfortunately.

You might think Kamailio is transmitting it to Asterisk, however
without looking at the actually routing tables on Kamailio you'll
never know if it actually made it to Asterisk.  Again, we need a pcap
trace on both Kamailio and Asterisk, plus what your routes look like
(route -n), for a call.  It will show us clearly what is happening.

This all sounds like a routing issue, so your network admins should be
able to help troubleshoot.

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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Paul Belanger]

On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
 On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham
 dcunning...@voisonics.com wrote:
 Hi Paul,

 Using ngrep/tcpdump shows the packet clearly going from the Kamailio server
 and arriving at the Asterisk server. This is why it's a mystery that
 Asterisk doesn't see the call coming in. We tried removing the firewall (so
 iptables -L shows no rules at all) but that didn't help unfortunately.

 You might think Kamailio is transmitting it to Asterisk, however
 without looking at the actually routing tables on Kamailio you'll
 never know if it actually made it to Asterisk.  Again, we need a pcap
 trace on both Kamailio and Asterisk, plus what your routes look like
 (route -n), for a call.  It will show us clearly what is happening.

 This all sounds like a routing issue, so your network admins should be
 able to help troubleshoot.

I finally re-read the complete thread.  When are you starting the VPN
on your Asterisk server, before or after Asterisk has started? If
after, and you are binding to 0.0.0.0, it is likely Asterisk is not
actually bound to your tun0 interface.  So, for a test, explicitly
have asterisk listen only on the tun0 interface, retry your call.

Or setup your tunnel, then stop Asterisk and start it again, that
should cause it to bind properly.


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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Paul Belanger]

On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham
dcunning...@voisonics.com wrote:
 On 22 January 2014 09:11, Steve Edwards asterisk@sedwards.com wrote:

 (Please don't top-post.)


 On Wed, 22 Jan 2014, David Cunningham wrote:

 We did send bindaddr to the VPN address and restarted Asterisk, but
 unfortunately that didn't solve the issue. Asterisk didn't complain, but
 still the sip set debug on didn't show the packets.


 Have you confirmed via 'netstat' (or some other system level toop) that
 Asterisk is actually listening to UDP port 5060 on the VPN IP address?

 --
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000


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 Hi Steve,

 When we have bindport = 172.x.x.14 then netstat -udpln shows the
 following. When bindport is 0.0.0.0 then netstat shows it listening on
 0.0.0.0 as you'd expect.

 udp0  0 172.x.x.14:50600.0.0.0:*
 18114/asterisk


 --
 David Cunningham, Voisonics
 http://voisonics.com/
 USA: +1 213 221 1092
 UK: +44 (0) 20 3298 1642
 Australia: +61 (0) 2 8063 9019

At this point in time, you'll need to show us a .pcap on the Asterisk
box, when you make a call to it via Kamailio.

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Re: [asterisk-users] Asterisk Fax detection *11.7

[Paul Belanger]

On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger ja...@j-mb.de wrote:
 Hello everybody

 I'm trying to enable the Digium res_fax app at my *11.7 Server.

 a fax show stats comes up with
 FAX Statistics:
 ---

 Current Sessions : 0
 Reserved Sessions: 0
 Transmit Attempts: 0
 Receive Attempts : 1
 Completed FAXes  : 1
 Failed FAXes : 1

 Digium G.711
 Licensed Channels: 1
 Max Concurrent   : 0
 Success  : 0
 Switched to T.38 : 0
 Canceled : 0
 No FAX   : 0
 Partial  : 0
 Negotiation Failed   : 0
 Train Failure: 0
 Protocol Error   : 0
 IO Partial   : 0
 IO Fail  : 0

 Digium T.38
 Licensed Channels: 1
 Max Concurrent   : 1
 Success  : 0
 Canceled : 0
 No FAX   : 0
 Partial  : 0
 Negotiation Failed   : 0
 Train Failure: 1
 Protocol Error   : 0
 IO Partial   : 0
 IO Fail  : 0

 so that should be ok.

 The corresponding dialplan section starts with


 [from-sip]
 include = inbound

 [inbound]
 exten = _X.,1,Answer()
 exten = _X.,n,GotoIf(${BLACKLIST()}?black,1)
 exten = _X.,n,Ringing
 exten = _X.,n,Progress()
 exten = _X.,n,Wait(5)
 exten = _X.,n,Dial(SIP/123SIP/456,30,oxX)
 ...
 exten = fax,1,NoOp( FAX DETECTED )
 exten = fax,n,Goto(fax-rx,receive,1)

 in the sip.conf i specified

 [general]
 sendrpid=rpid
 trustrpid=yes
 language=de
 videosupport=yes
 callevents=yes
 caninvite=yes
 qualify=yes
 nat=force_rport,comedia
 faxdetect=yes
 t38pt_udptl=yes

 ...

 [abcde]
 type=peer
 insecure=invite
 defaultuser=12345678912
 fromuser=12345678912
 fromdomain=abcde.ab
 secret=guess-what
 host=abcde.ab
 qualify=yes
 context=from-sip
 dtmfmode=rfc2833
 callbackextension=12345678912


 but all i can see if i try to send a testfax is

 == Using SIP VIDEO CoS mark 6
   == Using SIP RTP CoS mark 5
 -- Executing [12345678912@from-sip:1] Answer(SIP/abcde-0016, )
 in new stack
 0x7fd11404cd00 -- Probation passed - setting RTP source address to
 123.456.789.123:17108
 -- Executing [12345678912@from-sip:2] GotoIf(SIP/abcde-0016,
 0?black,1) in new stack
 -- Executing [12345678912@from-sip:3] Ringing(SIP/abcde-0016, )
 in new stack
 -- Executing [12345678912@from-sip:4] Progress(SIP/abcde-0016, )
 in new stack
 -- Executing [12345678912@from-sip:5] Wait(SIP/abcde-0016, 5) in
 new stack
 -- Executing [12345678912@from-sip:6] Dial(SIP/abcde-0016,
 SIP/123SIP/456,30,oxX) in new stack
   == Using SIP RTP CoS mark 5
   == Using SIP RTP CoS mark 5
 -- Called SIP/200
 -- Called SIP/201
 -- SIP/123-0018 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/456-0017 connected line has changed. Saving it until answer
 for SIP/abcde-0016
 -- SIP/123-0018 is ringing
 -- SIP/456-0017 is ringing


Don't expect T.30 over SIP to be reliable. If you need fax, you should
be using T.38. Your codec is likely the issue.

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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Paul Belanger]

On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham
dcunning...@voisonics.com wrote:
 Hi,

 We have a Kamailio and Asterisk cluster, both machines being on a real 103.x
 IP address and also on a 172.x OpenVPN address.

 The problem is that when Kamailo receives a call from the VPN and forwards
 it to the Asterisk server on it's 103.x address, Asterisk never sees the
 call.

 If Kamailio receives a call from the VPN and forwards the call to the
 Asterisk server on it's 172.x address then it works. However, if the call
 isn't from the VPN then forwarding it to the 172.x address doesn't work. So
 basically the problem is going between the real network and the VPN.

 The question is, how can we make this work when calls are received on either
 network on the Kamailio server and are forwarded to Asterisk?

 Using ngrep on the Asterisk server we see that it does receive the INVITE,
 but Asterisk's logging shows no sign it at all. We guess it's a Linux
 networking issue rather than Asterisk's fault, but don't know where to fix
 it. We do have net.ipv4.ip_forward = 1 on both the Kamailio and Asterisk
 servers.

 Thanks in advance for any help.

 The ngrep on the Asterisk server:

 U 2014/01/17 13:15:15.599557 172.x.x.x:5060 - 103.y.y.y:5060
 INVITE sip:9067268@103.y.y.y:5060;transport=udp SIP/2.0.
 Record-Route: sip:172.x.x.x;lr=on.
 Via: SIP/2.0/UDP 172.x.x.x;branch=z9hG4bK50c7.f49ceb73.0.
 Via: SIP/2.0/UDP 192.z.z.z:5062;rport=5062;branch=z9hG4bK806710997.
 From: 9067271 sip:9067271@172.x.x.x;tag=198791249.
 To: sip:9067268@172.x.x.x.
 Call-ID: 1905625787@192.z.z.z.
 ...

 172.x.x.x is the Kamailio server's VPN address
 103.y.y.y is the Asterisk server's real address
 192.z.z.z is the calling phone's LAN address

Sounds like a routing problem opposed to an application issue. You'll
have to fire up tcpdump on Kamailio and see what happens to the
packet. The look at the local routing tables to see where it is
getting routed.  If Asterisk is not receiving the patch, then Kamailio
is not routing it properly.

You'll be able to see everything once you have a pcap of the call.

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Re: [asterisk-users] Asterisk not receiving call from VPN address

[Paul Belanger]

On Mon, Jan 20, 2014 at 4:24 PM, David Cunningham
dcunning...@voisonics.com wrote:
 Hi Paul,

 The ngrep on the Asterisk server does show it being received. Have you any
 idea what would prevent it getting from the network stack to Asterisk on
 that machine?

Well, you need to use tcpdump on each hop across your network. If are
Asterisk is not getting anything, either it is not receiving anything
(check transmit side) or the firewall is dropping it.

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Re: [asterisk-users] How to install TEST_FRAMEWORK(E) ?

[Paul Belanger]

On 14-01-17 12:55 PM, Richard Mudgett wrote:

On Fri, Jan 17, 2014 at 3:42 AM, Olivier oza.4...@gmail.com wrote:


Hi,

I've installed a brand new Asterisk 12.0.0 system in which I can see, with
make menuselect, in Test Modules tab, that each test entry such as test_acl
can't be installed due a to missing TEST_FRAMEWORK(E) dependency.

Where this TEST_FRAMEWORK(E) comes from ?
How can it be installed ?
I've installed testsuite according
https://wiki.asterisk.org/wiki/display/AST/Installing+the+Asterisk+Test+Suitebut
 stopped before installing Third party modules (lua-devel and on).



TEST_FRAMEWORK is an option selectable under the Compiler Flags -
Development menu in menuselect.


./configure --enable-dev-mode

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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Paul Belanger]

On 14-01-16 03:37 PM, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as plan B).

As I never worked with H.323 channels in Asterisk earlier, I'm not sure if
it's stable enough to be used in production.

Googling about the subject didn't help much, I could only find some old and
probably outdated information which I don't want to rely on.

Can you please confirm if the OOH323 module in Asterisk 11 is stable enough
to use for voice calls? No extra functionality is needed, just to be able
to create a H.323 trunk towards the provider and make and receive a maximum
of 30 simultaneous voice calls through the trunk.

Thanks for your kind response!

Save yourself time / energy and insist using SIP. If your ITSP cannot 
accommodate your request, thank them and look for another provider.


H323 is Asterisk is basically dead, sure there is a module, sure it 
might compile, but you'll be going down the path of zero help.


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Re: [asterisk-users] Does cdr adaptive odbc re-connect automatically after a long idle time?

[Paul Belanger]

On Sat, Jan 11, 2014 at 4:56 PM, Charles Wang lazy.char...@gmail.com wrote:
 Hi all,

 I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc
 to write CDR to my MySQL's cdr table.
 After my testing, this scenario is working well.

 After a long idle time, I didn't make any call to the asterisk server.
 When I try to make a call again after 8 hours, I found that the cdr lost. It
 cannot be inserted to cdr table.
 Also, I could not find the insert CDR messages in the CLI at this period.

 Could you please tell me which settings are wrong? Why dose my odbc
 connection not re-connect to MySQL automatically?


 I checked the setting below:

 CLI:
 ubuntu*CLI cdr show status

 Call Detail Record (CDR) settings
 --
   Logging:Enabled
   Mode:   Simple
   Log unanswered calls:   Yes
   Log congestion: Yes

 * Registered Backends
   ---
 cdr-custom
 Adaptive ODBC
 csv

 ubuntu*CLI odbc show all

 ODBC DSN Settings
 -

   Name:   asterisk
   DSN:asterisk-connector
 Last connection attempt: 2014-01-11 18:16:40
   Pooled: Yes
   Limit:  1000
   Connections in use: 0


 -- /etc/asterisk/cdr.conf lists below:
 [general]
 enable=yes
 unanswered = yes
 congestion = yes
 endbeforehexten=yes

 [csv]
 usegmtime=no; log date/time in GMT.  Default is no
 loguniqueid=yes  ; log uniqueid.  Default is no
 loguserfield=yes ; log user field.  Default is no
 accountlogs=yes  ; create separate log file for each account code. Default
 is yes

 -- /etc/odbc.ini
 [asterisk-connector]
 Description   = MySQL connection to 'asterisk' database
 Driver= MySQL
 Database  = mydatabase
 Server= localhost
 UserName  = root
 Password  = mypassword
 Port  = 3306
 Socket= /var/run/mysqld/mysqld.sock


 -- /etc/asterisk/res_odbc.conf lists below:
 [ENV]

 [asterisk]
 enabled = yes
 dsn = asterisk-connector
 password = mypassword
 pre-connect = yes
 sanitysql = select 1
 pooling = yes
 idlecheck = 30
 share_connections = yes
 limit = 1000
 connect_timeout = 60
 negative_connection_cache = 600


 -- /etc/asterisk/cdr_adaptive_odbc.conf lists below:
 [cdr]
 connection=asterisk
 table=cdr
 alias start = calldate
 alias phoneno = phoneno
 alias userid = userid
 alias callerid = callerid

I would be inclined to check the database side over asterisk. We use
almost the same setup and don't have any issues. We go some time 12
hours between calls.  Once thing you could do is enable debug logs and
see what Asterisk is doing when the odbc connection is down.  EG: it
should be attempting to reconnect.

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Re: [asterisk-users] Allowing calls to m...@mydomain.org securely on Asterisk 11 box?

[Paul Belanger]

On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
 Hi all,

 I'm looking into adding the ability to call me at m...@mydomain.org on my
 Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
 this kind of access as securely as possible?

Well, if you want anybody to call you, you need to leave it open to
the public.  Meaning, you can't really secure it.  Obviously, don't
have any outbound trunks configured on the box so that the only
location some could dial would be your extension.

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Re: [asterisk-users] How to read IRQs and timing slips values

[Paul Belanger]

 into the software.

All that to say, try upgrading DAHDI and see what happens.

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Re: [asterisk-users] Dropped call on new CISCO router for no reason!

[Paul Belanger]

On 14-01-06 09:27 AM, Nick Cameo wrote:

Hello Everyone,

Just getting in a new cisco router, and would really like to get it up and
running as soon
as possible. Everything is configured from what we can see. This is a NAT
setup.
After 2 seconds on a successfully established call we reach retrans max,
and asterisk
disconnects the call. We have no idea why this is happening. SIP and RTP is
flowing as
expected.

Your help is greatly appreciated,

Nick.




Show us the problem, give us a SIP trace[1].

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

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Re: [asterisk-users] AppKonference 2.5

[Paul Albrecht]

On Dec 17, 2013, at 1:29 AM, virendra bhati virbh...@gmail.com wrote:

 Good Paul,
 
 I used Konference a lot very nice apps, but will this work with asterisk 
 latest version or not ?
 

It should work on the latest asterisk version. 

 I used asterisk 1.4,1.8 but didn't work on 11...
 
 
 On Mon, Dec 16, 2013 at 10:21 PM, Paul Albrecht palbre...@glccom.com wrote:
 Hi,
 
 I have released AppKonference 2.5 today.
 
 This release fixes a bug that can cause audio problems when conference frame 
 caching is enabled. It also fixes the spy feature so that more than one spyer 
 can spy on a channel at the same time. If more than one spyer is unmuted, 
 their audio is mixed and whispered to the spyee.
 
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 +91-9718500594
 +91-9250078532
 Asterisk Developer
 E-mail-: virbh...@gmail.com
 Skype id:- virbhati2
 New Delhi(India)
 
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[asterisk-users] AppKonference 2.5

[Paul Albrecht]

Hi,

I have released AppKonference 2.5 today.

This release fixes a bug that can cause audio problems when conference frame 
caching is enabled. It also fixes the spy feature so that more than one spyer 
can spy on a channel at the same time. If more than one spyer is unmuted, their 
audio is mixed and whispered to the spyee.

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Re: [asterisk-users] A Question about Management/Control Protocol Licensing

[Paul Belanger]

On 13-12-11 03:15 PM, Steve Murphy wrote:

I see the following paragraph in the Asterisk trunk LICENSE file:

In addition, Asterisk implements two management/control protocols: the
Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
(AGI). It is our belief that applications using these protocols to
manage or control an Asterisk instance do not have to be licensed
under the GPL or a compatible license, as we believe these protocols
do not create a 'derivative work' as referred to in the GPL. However,
should any court or other judiciary body find that these protocols do
fall under the terms of the GPL, then we hereby grant you a license to
use these protocols in combination with Asterisk in external
applications licensed under any license you wish.

This probably originated some years ago, and I wonder if Digium or the
Asterisk
community might consider adding the OTHER management/control protocols to
this
list: ARI, and the ExternalIVR interface.

If not, it might be instructive to learn why!

Would also like to see this update to include ARI. We talked a little 
about it at astridevcon, and I think it is likely an oversight.


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Re: [asterisk-users] Call Queue advise

[Paul Belanger]

On 13-12-09 06:47 PM, Bryan Anderson wrote:

I have a call queue that rings about 15 users and they are wanting to set
it up so that the last person to answer a call doesn't ring on the next
incoming call.

What would be the best way to handle this?  I have been looking at the
strategies and none of those seem to be right for this.  My current
thoughts are probably a macro that places a penalty on the user tell the
next call is answered.

Any advice for this would be greatly appreciated.

You have agents that log into a queue that don't want to get calls? Is 
that what you are saying?


Options 1 - log the agent out, they don't get the next call.

Option 2 - Set up weights for your agents, as answer a new call, 
increment then up so they don't get the next.


Either way, I see issues with the setup.  Best ways is to rethink your 
queue strategy and stop using ring all.


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Re: [asterisk-users] Asterisk on Windows

[Paul Belanger]

On 13-12-04 10:19 AM, CDR wrote:

Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework in Windows,
something simple to install, they could go public on the product.
Linux has a very steep learning curve. A Windows application that
would do exactly the same would be a home run. Note: I am a Linux
expert user, but it took me years to get here. And still, moving from
regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET
framework and Windows server 2012 are miles away in terms of
friendliness and on equal footing on performance. I don´t mean another
slow cygwin port, I man a native Asterisk for windows. In fact, I
would invest on the project if somebody wants to do it.


Do you just sit around and think shit up to blame Digium all day?

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Re: [asterisk-users] issue with speech in IVR

[Paul Belanger]

On 13-11-28 05:22 AM, Salaheddine Elharit wrote:

hello,

i have add the the code below but the issue still the same i can't go to
the project during the speech
any other solution

best regards

NB:for the version of asterisk i can't move to another version for the
moment

exten = _X,1,NoOp(Digit entered during prompt)
exten = _X,2,Goto(project,s,1)

Then you have a DTMF issue, Background will allow DTMF to interrupt the 
prompts.


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Re: [asterisk-users] Asterisk uses 105% CPU

[Paul Belanger]

] features.c
ast_features_init()
0x7f993c0fe700 tps_processing_function started at [  468]
taskprocessor.c ast_taskprocessor_get()
0x7f993c17a700 do_devstate_changes  started at [  750] devicestate.c
ast_device_state_engine_init()
0x7f993f17b700 logger_threadstarted at [ 1143] logger.c
init_logger()
0x7f993f1f7700 listener started at [ 1483] asterisk.c
ast_makesocket()
0x7f993f273700 tps_processing_function started at [  468]
taskprocessor.c ast_taskprocessor_get()
55 threads listed.

First thing, prune your Asterisk configuration and don't load any 
modules you don't need to use.  Are you really using chan_mgcp, 
chan_skinny, res_calender, etc.


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Re: [asterisk-users] issue with speech in IVR

[Paul Belanger]

On 13-11-27 04:57 PM, Salaheddine Elharit wrote:

hello list

i have an IVR menu in asterisk 1.4

like below

exten = 600,1,Ringing()
exten = 600,n,Wait(2)
exten = 600,n,Goto(home,s,1)




[home]
exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}music1)
exten = s,n,Background(${sounds_path}music2)
exten = s,n,Background(${sounds_path}music3)
exten = s,n,WaitExten(5)
exten = s,n,goto(home,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,WaitExten(5)
exten = i,n,goto(home,s,1)
exten = 1,1,Goto(project,s,1)


exten = _X,1,NoOp(Digit entered during prompt)
exten = _X,2,Goto(project,s,1)



[project]


exten = s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten = s,n,Background(${sounds_path}mymusic)
exten = s,n,WaitExten(5)
exten = s,n,Goto(project,s,1)
exten = i,1,Playback(${sounds_path}error)
exten = i,n,goto(project,s,1)

my problem when the customor call the number 600 and press 1 in order to go
to the project menu  he must wait all the speech music1 music2 and music 3

if there is any way to go to project menu during the speech

thanks and regards






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Re: [asterisk-users] Asterisk uses 105% CPU

[Paul Belanger]

On 13-11-27 07:35 AM, Jonas Kellens wrote:

Server specs :

XEON E3-1220V2
4 GB RAM
2 x 500GB HD (RAID0)
1 U
HOT-PLUG PSU

Linux sip.server.tld 2.6.32-358.18.1.el6.x86_64 #1 SMP Wed Aug 28
17:19:38 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux


There is no transcoding. Calls are using G711a.

Maybe there is some trancoding when using voicemail...

How can I find out if there is trancoding ??

Maybe explain what your dialplan is doing. Are you making system calls 
to a database or AGI?


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Re: [asterisk-users] CEL for attented transfer

[Paul Belanger]

On 13-11-19 11:03 AM, Jean-Denis Girard wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hi Jairo,

Le 19/11/2013 01:36, Jairo a écrit :

https://wiki.asterisk.org/wiki/display/AST/CEL+Events+and+Fields


Thanks for your reply, but I have read this page of the wiki, I know
what the fields mean.

Well, it is a way lot harder to figure out because you used 
features.conf.  Because of this, local channels are involved.



What I don't understand is how the events in my example can be used to
determine 107 was attended transferred to 103 by 100.

Or I do know that Local/103@100-0042;1 and Local/103@100-0042;2 were
created by asterisk when SIP/100-0275 asked for atxfer?

How does the event ATTENDEDTRANSFER/ SIP/107-0274/ Local/103@100-0042;1
show that 107 is transferred to 103?

Specifically, you are going to have to track the channel IDs and look at 
the sequence of events.  Then make an educated guess about what is 
happening.


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Re: [asterisk-users] calendar.conf include

[Paul Belanger]

On 13-11-13 10:20 AM, Jonas Kellens wrote:

Hello,

can I use include-statements in the calendar.conf configuration file ?



You _should_ be able to use it will every .conf file, otherwise it is a bug.

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Re: [asterisk-users] Unix connections not always disconnecting

[Paul Belanger]

On 13-11-07 10:31 AM, Ishfaq Malik wrote:

On 7 November 2013 15:26, Gareth Blades mailinglist+aster...@dns99.co.ukwrote:


On 07/11/13 11:20, Ishfaq Malik wrote:


Hi

We are using asterisk 1.8.23.1

We have a script that runs on a minute cron which polls the asterisk
server for 3 bits of information by using

asterisk -rx 'command'

which then gets pushed to a graphite server we have

99% of this runs smoothly.




Out of interest what are you trying to monitor?

We tend to use cacti for graphing and snmp provides all the information we
require.




Active calls, sip peers connected, sip peers disconnected and then breaking
all of those down by customer as we run a multi tenanted set up.

SNMP would give us totals but I don't think it would do the breakdown by
customer.

You should avoid using the CLI to access that information. You'd likely 
getter better results using AMI or CEL.


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Re: [asterisk-users] Capture dead phone?

[Paul Belanger]

On 13-11-07 07:51 PM, Mitch Claborn wrote:

Asterisk 11.1

Is it possible to catch the fact that an IP phone has died in the middle
of a call and do something with it in the dialplan?

Background: we run a small call center.  Our agents sit in two groups,
with their IP phones running from 2 different switches. Every once in a
while the power on one side of the room will go out, or one of the
switches will die, or one of the agents will knock something loose with
their foot.  If/when that happens while the agent is on a call with a
customer, I'd like to be able to save that caller and put them back in
the queue (at the head of the queue).

No, you won't be able to save the call if the far end goes down.  Best 
you could do would be to enable qualify, track then the agent phone goes 
offline, if a call also drop around that time frame, initial some sort 
of callback.


However, solve the issue at the source. Spend the money for a UPS at 
each desktop, convert your phones to PoE and install a UPS in your 
server room.


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Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Paul Belanger]

On 13-10-28 06:03 PM, Patrick Lists wrote:

On 10/28/2013 07:29 PM, Eddie Mikell wrote:

All,

The users in our organization are well, quite frankly, sick of phone
service that is being provided.  The choppy phone calls, and drop outs
are detrimental to our sales force.

I've tried about everything I can think of.

Moved the asterisk server from VM machine to dedicated machine


That's a good start. Now what have you done to conclude that the
Asterisk server is not the cause of your problems?


More than enough bandwidth


That's irrelevant. It's about the quality of that bandwidth. Have you
figured out if there might be a lot of packetloss or are you perhaps on
a cablelink which is a *shared* medium? Once your link hits the box in
the street it shares it with others who might be eating up all the
bandwidth with their torrent downloads etc.? Use tools like iperf, smoke
ping and mtr to see if there are obvious problems on the route to your
VoIP provider.


Setting 802.1p = 7

Set Dedicated voice traffic 35% of bandwidth.

Not sure what option would be the best


Once the packets leave your premises and your ISP/cable company starts
messing with them a QoS setting is generally not honored so not very
helpful unless your LAN is congested.


Put analog lines in the conference room to avoid the dropouts -
leave the sip lines in place for day to day use


If those analog lines are cheap, easy to get then as an intermediate
solution I would order those analog lines as fast as I could. Or fix the
VoIP problems, whichever is faster.


Hire a consultant


An experienced VoIP consultant should be able to tell you what is or
could be causing your problems. With your users sick of phone service
it suprises me that you haven't already hired one.


Ditch the system and buy a pre-packaged system - RingCentral or some
such.


And what if it's your Internet link or the route to your VoIP provider?
What if your VoIP provider is messing up?


There are no local asterisk professionals who can help, and we are a
little leery of opening up our system to outside consultants.


If you don't want that then you don't want that but given the state your
users are in I would be less worried about giving a Consultant access to
the Asterisk box and more worried about my job :-)


Anyone else face the above, and finally abandoned Asterisk for a
commercial system?


I have seen that once years ago where some clueless sales guy had
totally oversold an ancient Asterisk/Bristuff/ISDN setup which was very
buggy and crash prone. There was no way to make that work reliably.
After the supplier failed for months I was brought in to review the
setup and possibly fix it. Told the customer to cut its losses. So they
kicked out their supplier and opted for a different setup.


We have 167 users.
I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
conference rooms.


I don't know how Grandstream is these days. I thought the GXP2100 was ok
but I guess you already know if there's a problem with those phones from
the (lack of) intra-office call complaints from your users.


Suggestions welcome.


Hire a Consultant or someone who has been part of this Community for a
while and is well known on this list or in #asterisk on irc. Provide
remote access if required. Change passwords afterwards.

If you really don't want to provide remote access then find a reputable
VoIP provider with a switch physically as close as possible to your
location, get a DID for a few bucks, hook it up to your Asterisk box and
route it to a line on your phone, grab your cell, call that DID and see
if you still have the problem. It wouldn't be the first time that the
link between you and your VoIP provider just doesn't cut it. Or maybe
your VoIP provider just sucks and you need to change to a different one.
Both flowroute.com and voip.ms work well for me (no affiliation). Or
maybe your Internet link sucks and you need to change your ISP.


^ this

Like others said, you really need to drill down and find out where your 
audio issues are. Local is easy to do, since you control the network, 
remote is harder.


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Re: [asterisk-users] Is this big of new modification in Asterisk Events Objects values ?

[Paul Belanger]

On 13-10-25 07:06 AM, virendra bhati wrote:

Hi Team,

Thanks for your great job an Asterisk new features developments. I
installed asterisk-12 Beta and found some changes as well which i notice to
put in-front of your knowledge, don't know that bug of new modification
into objects or old version (asterisk-11) mistake corrected that time,
anyway

*Asterisk-12:*

Array
(
[Event] = ConfbridgeMute
[Privilege] = call,all
[Conference] = 42
[BridgeUniqueid] = 9f2ae5df-0749-4494-b8b7-12eb50dc765d
[BridgeType] = base
[BridgeTechnology] = softmix
[BridgeNumChannels] = 2
[Channel] = SIP/5000-0006
[ChannelState] = 6
[ChannelStateDesc] = Up
*[CallerIDNum] = 5000*
[CallerIDName] = 5000
[ConnectedLineNum] = unknown
[ConnectedLineName] = unknown
[AccountCode] =
[Context] = from-sip
[Exten] = 1234
[Priority] = 3
[Uniqueid] = 1382599433.22
)

Please check the BOLD section. earlier is was  *[CallerIDnum]

*
*So 'n' is now 'N'
*

Asterisk AMI got basically a rewrite[1] of how it works, so there are 
some breaking changes moving forward.


Read ChangeLog and UPGRADE.txt in the source tree for more information.


[1] https://wiki.asterisk.org/wiki/display/AST/AMI+1.4+Specification




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