[asterisk-users] Asterisk and XMPP - AstriCon 2016

2016-09-30 Thread Marcelo Terres 
Hello.

This is the link of the slides of my talk presented yesterday in
AstriCon, about Asterisk and XMPP.

As soon as the video is available, I'll share it too.

http://pt.slideshare.net/mhterres/astricon-2016-using-asterisk-and-xmpp-to-provide-greater-tools-to-your-customers-and-your-users

[]s

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

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[asterisk-users] Asterisk 14.0.2 Now Available

2016-09-30 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 14.0.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 14.0.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

Bugs fixed in this release:
---
 * ASTERISK-26410 - core: Asterisk 14 doesn't show the header in
  the console or verbose when starting (Reported by Dan Jenkins)
 * ASTERISK-26426 - format_ogg_opus: remove from source (Reported
  by Kevin Harwell)
 * ASTERISK-26425 - download_externals: ignore xmlstarlet return
  code for optional element (Reported by Kevin Harwell)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-14.0.2

Thank you for your continued support of Asterisk!

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[asterisk-users] asterisk security framework

2016-09-30 Thread marek cervenka 

hi,

i'm trying configure $subj

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger

but there is a ton of "informational" messages

[Sep 30 14:40:16] SECURITY[18311] res_security_log.c: 
SecurityEvent="SuccessfulAuth",EventTV="2016-09-30T14:40:16.833+0200",Severity="Informational",Service="PJSIP",EventVersion="1",AccountID="webrtc",SessionID="e74a4f71-4f0a-14ee-6373-a53d0540dd12",LocalAddress="IPV4/WSS/1.1.1.1/39512",RemoteAddress="IPV4/WSS/1.1.1.1/39512",UsingPassword="1"


is there possibility log only "important" things?

i.e. by severity or by category from 
https://wiki.asterisk.org/wiki/display/AST/Security+Events+to+Log


thanks

Marek



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Re: [asterisk-users] Hello again

2016-09-30 Thread A J Stiles 
On Friday 30 Sep 2016, aaberga/gmail wrote:
> Hi,
> 
> after a long pause (Asterisk 1.8 times), I have started again playing with
> VOIP. A lot has changed since last time I did setup an Asterisk system!
> 
> So I am asking for some help.
[stuff deleted]
> [2102]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2102
> aors=2102
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no

You might want to comment out all references to g729  (which needs a special 
licence)  and just use alaw  (the native codec of the PSTN)  throughout.

If one of the phones is deciding to use g729 and your Asterisk doesn't have 
the relevant licence, then you might well get all manner of strange things 
happening.

Even if you have g729 licences, try and get it working with alaw first.  The 
fewer things there are that could go wrong, the better.  It's always best to 
get it working with the simplest possible setup first, and only then add 
sophistication.

-- 
AJS

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

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Re: [asterisk-users] Hello again

2016-09-30 Thread Joan Aymà [ackstorm] 
Your external addres seems wrong. As you are doing natting, you need to
set to you external (natted behind firewall).


El 30/09/16 a les 13:07, aaberga/gmail ha escrit:
> Hi,
>
> after a long pause (Asterisk 1.8 times), I have started again playing with 
> VOIP. A lot has changed since last time I did setup an Asterisk system!
>
> So I am asking for some help.
>
> 
>
> PJSIP seems tougher..
>
> So my problem is that I do have a test system up in the cloud, behind a 
> firewall. I am trying to make the “Hello World!” mandatory call between two 
> iPhones (with the Bria SIP client).
>
> Outcomes are erratic.
>
> 
>
> This is the pjsip.conf file:
>
> ——
>
> [transport-udp-nat]
> type=transport
> protocol=udp
> bind=0.0.0.0
> local_net=10.2.12.3/32
> local_net=127.0.0.1/32
> external_media_address=10.2.12.2
> external_signaling_address=10.2.12.2
>
> ;===Messagenet TRUNK 
>
> [messagenet_reg]
> type=registration
> transport=transport-udp-nat
> outbound_auth=messagenet_auth
> server_uri=sip:xx...@sip.messagenet.it:5061
> client_uri=sip:xx...@sip.messagenet.it:5061
>  
> [messagenet_auth]
> type=auth
> auth_type=userpass
> password=
> username=
>  
> [messagenet_aor]
> type=aor
> contact=sip:sip.messagenet.it:5061
>  
> [messagenet]
> type=endpoint
> transport=transport-udp-nat
> context=messagenet_incoming
> disallow=all
> allow=ulaw
> allow=alaw
> outbound_auth=messagenet_auth
> aors=messagenet_aor
>  
> [messagenet_id]
> type=identify
> endpoint=messagenet
> match=sip.messagenet.it
>  
> ;===Extension 2102
>
> [2102]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2102
> aors=2102
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no
>
>
> [auth2102]
> type=auth
> auth_type=userpass
> password=xx
> username=2102
>  
> [2102]
> type=aor
> max_contacts=1
>  
> ;===Extension 2103
>
> [2103]
> type=endpoint
> context=internal
> ;disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> transport=transport-udp-nat
> auth=auth2103
> aors=2103
> rtp_symmetric=yes
> force_rport=yes
> ice_support=yes
> direct_media=no
>  
> [auth2103]
> type=auth
> auth_type=userpass
> password=xx
> username=2103
>  
> [2103]
> type=aor
> max_contacts=1
>  
> 
>
> This is a trace of what I do see from the console.
>
> First I let the Bria clients connect. Then I try to call terminal 1 from 
> terminal 2. Most of the times there is no route to the destination, even if 
> it appears as an online AOR (whatever that means!! Ahhh: Good olde times of 
> Peer, Friend, etc… ;-)
>
> A couple of times I got a connection, with the typical one side only audio of 
> NAT traversal problems.
>
> BTW: The iPhones are behind TWO nats (one is given by the broadband router, 
> one by the WiFi router that gives a better WiFi cover for in-house things).
>
> My understanding is that I did something wrong in letting the phones 
> ‘register’ them as present and available to receive calls. 
>
> If only I knew what is wrong… I have tried random combinations of 
> rtp_symmetric, force_rport, and friends; nothing final discovered...
>
> 
>
> Thanks in advance for any help,
> Aldo
>
>
> PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP 
> side. The only catch is that Zoiper has less than optimal background support 
> on IOS… And I have no plan to make an IAX client myself!
>
> I want to get my old Asterisk apps back online and the VOIP client part makes 
> no sense to me..
>
>

-- 
Joan Aymà
Departamento de SAT
joan.a...@ackstorm.es 

ackstorm 
 
C/ Catalunya 72 Local, 08840 Viladecans - Barcelona
902 888 345  | i...@ackstorm.es
 | **ackstorm.es** 
 
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[asterisk-users] Hello Again - ooops

2016-09-30 Thread aaberga/gmail 
Sorry forgot to attach the CLI trace:

=

CLI> pjsip show aors

  Aor:
Contact:  

 
=

  Aor:  210220

  Aor:  210320

  Aor:  messagenet_aor   0
Contact:  messagenet_aor/sip:sip.messagenet.it:5061  Unknown
   nan


-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=eabdf84e26104b07' from AOR '2103' due 
to request
-- Added contact 'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' 
to AOR '2103' with expiration of 900 seconds
-- Removed contact 
'sip:2103@192.168.155.5:63639;rinstance=420fa67d404d9816' from AOR '2103' due 
to request
-- Added contact 'sip:2103@37.228.255.229:60677;rinstance=635ece4650faa34e' 
to AOR '2103' with expiration of 900 seconds
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=0833518dac88d43b' from AOR '2102' due 
to request
-- Added contact 'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' 
to AOR '2102' with expiration of 900 seconds
-- Removed contact 
'sip:2102@192.168.155.5:60157;rinstance=917b91dc461a6eda' from AOR '2102' due 
to request
-- Added contact 'sip:2102@37.228.255.229:60605;rinstance=fbd37b6a6d7cb4fb' 
to AOR '2102' with expiration of 900 seconds


-- Executing [2102@internal:1] Set("PJSIP/2103-0003", "ORIGIN=IP") in 
new stack
-- Executing [2102@internal:2] NoOp("PJSIP/2103-0003", "Declared 
CallerID=<"2103" <2103>>") in new stack
-- Executing [2102@internal:3] Set("PJSIP/2103-0003", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2102@internal:4] Set("PJSIP/2103-0003", 
"OriginalEXTEN=2102") in new stack
-- Executing [2102@internal:5] Set("PJSIP/2103-0003", 
"CDR(userfield)=2102") in new stack
-- Executing [2102@internal:6] Goto("PJSIP/2103-0003", 
"dialplan-switch,2102,1") in new stack
-- Goto (dialplan-switch,2102,1)
-- Executing [2102@dialplan-switch:1] NoOp("PJSIP/2103-0003", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2102@dialplan-switch:2] Dial("PJSIP/2103-0003", 
"PJSIP/2102") in new stack
[Sep 30 10:50:44] ERROR[19237]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2102'
[Sep 30 10:50:44] ERROR[19237]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2102'
[Sep 30 10:50:44] WARNING[19287][C-0003]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2102@dialplan-switch:3] Hangup("PJSIP/2103-0003", "") in 
new stack
  == Spawn extension (dialplan-switch, 2102, 3) exited non-zero on 
'PJSIP/2103-0003'
  
  
-- Executing [2103@internal:1] Set("PJSIP/2102-0004", "ORIGIN=IP") in 
new stack
-- Executing [2103@internal:2] NoOp("PJSIP/2102-0004", "Declared 
CallerID=<"2102" <2102>>") in new stack
-- Executing [2103@internal:3] Set("PJSIP/2102-0004", 
"CALLERID(name)=Insicure-IP") in new stack
-- Executing [2103@internal:4] Set("PJSIP/2102-0004", 
"OriginalEXTEN=2103") in new stack
-- Executing [2103@internal:5] Set("PJSIP/2102-0004", 
"CDR(userfield)=2103") in new stack
-- Executing [2103@internal:6] Goto("PJSIP/2102-0004", 
"dialplan-switch,2103,1") in new stack
-- Goto (dialplan-switch,2103,1)
-- Executing [2103@dialplan-switch:1] NoOp("PJSIP/2102-0004", " 
Entering Dialplan Switch from  ") in new stack
-- Executing [2103@dialplan-switch:2] Dial("PJSIP/2102-0004", 
"PJSIP/2103") in new stack
[Sep 30 10:52:01] ERROR[19299]: res_pjsip.c:2106 
sip_get_tpselector_from_endpoint: Unable to retrieve PJSIP transport 
'transport-udp-nat' for endpoint '2103'
[Sep 30 10:52:01] ERROR[19299]: chan_pjsip.c:1788 request: Failed to create 
outgoing session to endpoint '2103'
[Sep 30 10:52:01] WARNING[19306][C-0004]: app_dial.c:2431 dial_exec_full: 
Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2103@dialplan-switch:3] Hangup("PJSIP/2102-0004", "") in 
new stack
  == Spawn extension (dialplan-switch, 2103, 3) exited non-zero on 
'PJSIP/2102-0004'


=



Tnx,
Aldo


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[asterisk-users] Hello again

2016-09-30 Thread aaberga/gmail 
Hi,

after a long pause (Asterisk 1.8 times), I have started again playing with 
VOIP. A lot has changed since last time I did setup an Asterisk system!

So I am asking for some help.



PJSIP seems tougher..

So my problem is that I do have a test system up in the cloud, behind a 
firewall. I am trying to make the “Hello World!” mandatory call between two 
iPhones (with the Bria SIP client).

Outcomes are erratic.



This is the pjsip.conf file:

——

[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=10.2.12.3/32
local_net=127.0.0.1/32
external_media_address=10.2.12.2
external_signaling_address=10.2.12.2

;===Messagenet TRUNK 

[messagenet_reg]
type=registration
transport=transport-udp-nat
outbound_auth=messagenet_auth
server_uri=sip:xx...@sip.messagenet.it:5061
client_uri=sip:xx...@sip.messagenet.it:5061
 
[messagenet_auth]
type=auth
auth_type=userpass
password=
username=
 
[messagenet_aor]
type=aor
contact=sip:sip.messagenet.it:5061
 
[messagenet]
type=endpoint
transport=transport-udp-nat
context=messagenet_incoming
disallow=all
allow=ulaw
allow=alaw
outbound_auth=messagenet_auth
aors=messagenet_aor
 
[messagenet_id]
type=identify
endpoint=messagenet
match=sip.messagenet.it
 
;===Extension 2102

[2102]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2102
aors=2102
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no


[auth2102]
type=auth
auth_type=userpass
password=xx
username=2102
 
[2102]
type=aor
max_contacts=1
 
;===Extension 2103

[2103]
type=endpoint
context=internal
;disallow=all
allow=ulaw
allow=alaw
allow=g729
transport=transport-udp-nat
auth=auth2103
aors=2103
rtp_symmetric=yes
force_rport=yes
ice_support=yes
direct_media=no
 
[auth2103]
type=auth
auth_type=userpass
password=xx
username=2103
 
[2103]
type=aor
max_contacts=1
 


This is a trace of what I do see from the console.

First I let the Bria clients connect. Then I try to call terminal 1 from 
terminal 2. Most of the times there is no route to the destination, even if it 
appears as an online AOR (whatever that means!! Ahhh: Good olde times of Peer, 
Friend, etc… ;-)

A couple of times I got a connection, with the typical one side only audio of 
NAT traversal problems.

BTW: The iPhones are behind TWO nats (one is given by the broadband router, one 
by the WiFi router that gives a better WiFi cover for in-house things).

My understanding is that I did something wrong in letting the phones ‘register’ 
them as present and available to receive calls. 

If only I knew what is wrong… I have tried random combinations of 
rtp_symmetric, force_rport, and friends; nothing final discovered...



Thanks in advance for any help,
Aldo


PS: Setting up Zoiper as IAX client works of course like a charm on the VOIP 
side. The only catch is that Zoiper has less than optimal background support on 
IOS… And I have no plan to make an IAX client myself!

I want to get my old Asterisk apps back online and the VOIP client part makes 
no sense to me..


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  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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