[asterisk-users] Unable to create Jingle session

2014-07-10 Thread Control Oye 
Dear All,

I have different Asterisk Servers most of them are version 1.8 - I have
recently upgrade to Asterisk version 11 on 2 servers.

I have Jabber ( chan_gtalk ) configured on 1.8 version  and it is working
within all 1.8 version servers.

I have XMPP ( chan_motif )  configured on 11 version and it is working with
all 11 versions servers.

When I try to call from version 11 ( usiing xmpp - chan_motif ) to version
1.8 server it is always failed with following error

==

-- Executing [872@longdistance:1] Dial("SIP/804-003d", "Motif/jitsi/
myfaken...@jit.si") in new stack

[Jul  5 16:52:21] ERROR[28270][C-002b]: chan_motif.c:1814
jingle_request: Unable to create Jingle session on endpoint 'jitsi' as no
capable resource for target 'myfaken...@jit.si' was found

[Jul  5 16:52:21] WARNING[28270][C-002b]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'Motif' (cause 42 -
Switching equipment congestion)

  == Everyone is busy/congested at this time (1:0/0/1)

==

Any solution - Please Help

Abdullah Faheem
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Need a developer to write me a patch

2014-07-10 Thread Rusty Newton 
On Thu, Jul 10, 2014 at 9:08 AM, CDR  wrote:
> I cannot wait for the regular bug-patch process to play out. I am
> considering  hiring a developer to fix bug 24015, and of course submit the
> patch for the bug. The issue is simple, the app Transfer does not transfer
> when using PJSIP.. I called Digium and they said that they do not do this
> kind of work.

The Asterisk users list is not intended for this sort of post. You can
post on the asterisk-biz list if you are posting a job offering or
looking for consultants.

Otherwise bug bounties are accommodated on the dev list:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 11.11.0 Now Available

2014-07-10 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 11.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.11.0 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-22551 - Session timer : UAS (Asterisk) starts counting
  at Invite, UAC starts counting at 200 OK. (Reported by i2045)
 * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
  by Peter Whisker)
 * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
  (Reported by Walter Doekes)
 * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
  categories but the requested one (Reported by zvision)
 * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
  results in several bridges with same conf_name (Reported by
  Iñaki Cívico)
 * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
  AMI when waiting to enter a conference (Reported by Matt Jordan)
 * ASTERISK-23683 - #includes - wildcard character in a path more
  than one directory deep - results in no config parsing on module
  reload (Reported by tootai)
 * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
  (Reported by Corey Farrell)
 * ASTERISK-23609 - Security: AMI action MixMonitor allows
  arbitrary programs to be run (Reported by Corey Farrell)
 * ASTERISK-23673 - Security: DOS by consuming the number of
  allowed HTTP connections. (Reported by Richard Mudgett)
 * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
  a DEBUG level of zero (Reported by Rusty Newton)
 * ASTERISK-23766 - [patch] Specify timeout for database write in
  SQLite (Reported by Igor Goncharovsky)
 * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
  with Lua 5.2 or greater due to addition of goto statement
  (Reported by Rusty Newton)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
  loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
  length if ICE (Reported by Richard Kenner)
 * ASTERISK-23790 - [patch] - SIP From headers longer than 256
  characters result in dropped call and 'No closing bracket'
  warnings. (Reported by uniken1)
 * ASTERISK-23917 - res_http_websocket: Delay in client processing
  large streams of data causes disconnect and stuck socket
  (Reported by Matt Jordan)
 * ASTERISK-23908 - [patch]When using FEC error correction,
  asterisk tries considers negative sequence numbers as missing
  (Reported by Torrey Searle)
 * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
  files  (Reported by Corey Farrell)
 * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
  objects that were already freed (Reported by Corey Farrell)
 * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
  between attributes (Reported by Alexander Traud)
 * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
  (Reported by Steve Davies)
 * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
  PI) in revision 413765 breaks working environments (Reported by
  Pavel Troller)

Improvements made in this release:
---
 * ASTERISK-23492 - Add option to safe_asterisk to disable
  backgrounding (Reported by Walter Doekes)
 * ASTERISK-22961 - [patch] DTLS-SRTP not working with SHA-256
  (Reported by Jay Jideliov)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.11.0

Thank you for your continued support of Asterisk!

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 12.4.0 Now Available

2014-07-10 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 12.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 12.4.0 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-22551 - Session timer : UAS (Asterisk) starts counting
  at Invite, UAC starts counting at 200 OK. (Reported by i2045)
 * ASTERISK-23792 - Mutex left locked in chan_unistim.c (Reported
  by Peter Whisker)
 * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
  (Reported by Walter Doekes)
 * ASTERISK-23499 - app_agent_pool: Interval hook prevents channel
  from being hung up (Reported by Matt Jordan)
 * ASTERISK-23721 - Calls to PJSIP endpoints with video enabled
  result in leaked RTP ports (Reported by cervajs)
 * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
  categories but the requested one (Reported by zvision)
 * ASTERISK-23718 - res_pjsip_incoming_blind_request: crash with
  NULL session channel (Reported by Jonathan Rose)
 * ASTERISK-23541 - Asterisk 12.1.0 Not respecting directmedia=no
  and issuing REINVITE (Reported by Justin E)
 * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
  results in several bridges with same conf_name (Reported by
  Iñaki Cívico)
 * ASTERISK-23824 - ConfBridge: Users cannot be muted via CLI or
  AMI when waiting to enter a conference (Reported by Matt Jordan)
 * ASTERISK-23683 - #includes - wildcard character in a path more
  than one directory deep - results in no config parsing on module
  reload (Reported by tootai)
 * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
  (Reported by Corey Farrell)
 * ASTERISK-21965 - [patch] Bug-fixed version of safe_asterisk not
  installed over old version (Reported by Jeremy Kister)
 * ASTERISK-23802 - Security: Deadlock in res_pjsip_pubsub on
  transaction timeout (Reported by Mark Michelson)
 * ASTERISK-23489 - Vulnerability in res_pjsip_pubsub:
  unauthenticated remote crash in during MWI unsubscribe without
  being subscribed (Reported by John Bigelow)
 * ASTERISK-23609 - Security: AMI action MixMonitor allows
  arbitrary programs to be run (Reported by Corey Farrell)
 * ASTERISK-23673 - Security: DOS by consuming the number of
  allowed HTTP connections. (Reported by Richard Mudgett)
 * ASTERISK-23766 - [patch] Specify timeout for database write in
  SQLite (Reported by Igor Goncharovsky)
 * ASTERISK-23844 - Load of pbx_lua fails on sample extensions.lua
  with Lua 5.2 or greater due to addition of goto statement
  (Reported by Rusty Newton)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
  loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23834 - res_rtp_asterisk debug message gives wrong
  length if ICE (Reported by Richard Kenner)
 * ASTERISK-23922 - ao2_container nodes are inconsistent REF_DEBUG
  (Reported by Corey Farrell)
 * ASTERISK-23790 - [patch] - SIP From headers longer than 256
  characters result in dropped call and 'No closing bracket'
  warnings. (Reported by uniken1)
 * ASTERISK-23917 - res_http_websocket: Delay in client processing
  large streams of data causes disconnect and stuck socket
  (Reported by Matt Jordan)
 * ASTERISK-23908 - [patch]When using FEC error correction,
  asterisk tries considers negative sequence numbers as missing
  (Reported by Torrey Searle)
 * ASTERISK-23947 - ActionID missing from AMI PJSIP events
  (PJSIPShowEndpoints, etc.) (Reported by Mark Michelson)
 * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
  files  (Reported by Corey Farrell)
 * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
  objects that were already freed (Reported by Corey Farrell)
 * ASTERISK-23916 - [patch]SIP/SDP fmtp line may include whitespace
  between attributes (Reported by Alexander Traud)
 * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
  (Reported by Steve Davies)
 * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
  PI) in revision 413765 breaks working environments (Reported by
  Pavel Troller)
 * ASTERISK-24001 - res_rtp_asterisk fails to load module due to
  undefined symbol 'dtls_perform_handshake' when PJPROJECT is not
  installed (Reported by Don Fanning)

Improvements made in this release:
---
 * ASTERISK-23492 - Add option to safe_asterisk to disable
  backgrounding (Reported by Walter Doekes)
 * ASTERISK-23654 - Add 'pjsip reload' to default cli_aliases.conf
  (Reported by Rusty Newton)
 * ASTERISK-23811 - Improve performance of Asterisk by reducing the
  number of channel snaps

[asterisk-users] Asterisk 1.8.29.0 Now Available

2014-07-10 Thread Asterisk Development Team 
The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 1.8.29.0 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-22551 - Session timer : UAS (Asterisk) starts counting
  at Invite, UAC starts counting at 200 OK. (Reported by i2045)
 * ASTERISK-23582 - [patch]Inconsistent column length in *odbc
  (Reported by Walter Doekes)
 * ASTERISK-23803 - AMI action UpdateConfig EmptyCat clears all
  categories but the requested one (Reported by zvision)
 * ASTERISK-23035 - ConfBridge with name longer than max (32 chars)
  results in several bridges with same conf_name (Reported by
  Iñaki Cívico)
 * ASTERISK-23683 - #includes - wildcard character in a path more
  than one directory deep - results in no config parsing on module
  reload (Reported by tootai)
 * ASTERISK-23827 - autoservice thread doesn't exit at shutdown
  (Reported by Corey Farrell)
 * ASTERISK-23814 - No call started after peer dialed (Reported by
  Igor Goncharovsky)
 * ASTERISK-23673 - Security: DOS by consuming the number of
  allowed HTTP connections. (Reported by Richard Mudgett)
 * ASTERISK-23246 - DEBUG messages in sdp_crypto.c display despite
  a DEBUG level of zero (Reported by Rusty Newton)
 * ASTERISK-23766 - [patch] Specify timeout for database write in
  SQLite (Reported by Igor Goncharovsky)
 * ASTERISK-23818 - PBX_Lua: after asterisk startup module is
  loaded, but dialplan not available (Reported by Dennis Guse)
 * ASTERISK-23667 - features.conf.sample is unclear as to which
  options can or cannot be set in the general section (Reported by
  David Brillert)
 * ASTERISK-23790 - [patch] - SIP From headers longer than 256
  characters result in dropped call and 'No closing bracket'
  warnings. (Reported by uniken1)
 * ASTERISK-23908 - [patch]When using FEC error correction,
  asterisk tries considers negative sequence numbers as missing
  (Reported by Torrey Searle)
 * ASTERISK-23921 - refcounter.py uses excessive ram for large refs
  files  (Reported by Corey Farrell)
 * ASTERISK-23948 - REF_DEBUG fails to record ao2_ref against
  objects that were already freed (Reported by Corey Farrell)
 * ASTERISK-23984 - Infinite loop possible in ast_careful_fwrite()
  (Reported by Steve Davies)
 * ASTERISK-23897 - [patch]Change in SETUP ACK handling (checking
  PI) in revision 413765 breaks working environments (Reported by
  Pavel Troller)

Improvements made in this release:
---
 * ASTERISK-23564 - [patch]TLS/SRTP status of channel not currently
  available in a CLI command (Reported by Patrick Laimbock)
 * ASTERISK-23492 - Add option to safe_asterisk to disable
  backgrounding (Reported by Walter Doekes)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.29.0

Thank you for your continued support of Asterisk!

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] CDR(dst) in AEL macro

2014-07-10 Thread Rafael dos Santos Saraiva 
Hi

I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field  CDR(dst), showing only ~~s~~.

I tried various configurations, but without solutions.

This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} > 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}

Thank's.

Att,
*Rafael dos Santos Saraiva*

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Need a developer to write me a patch

2014-07-10 Thread CDR 
I cannot wait for the regular bug-patch process to play out. I am
considering  hiring a developer to fix bug 24015, and of course submit the
patch for the bug. The issue is simple, the app Transfer does not transfer
when using PJSIP.. I called Digium and they said that they do not do this
kind of work.
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] busy() not setting PRI_CAUSE

2014-07-10 Thread Eric Wieling 
I would have to read the source code to know for sure.   Is it too much trouble 
to try my suggestion?


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 8:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] busy() not setting PRI_CAUSE

The description of busy() in the asterisk documentation wiki states:

"This application will indicate the busy condition to the calling channel."

Wouldn't 'indicate the busy condition' on a PRI channel imply setting cause 17?

-Justin

From: 
asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, July 09, 2014 4:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] busy() not setting PRI_CAUSE

Generally if you want to send a cause 17 to the caller you would use Hangup(17) 
and let the caller's switch generate the busy tone.

If the dialplan has already answered the call, then you might want to use Busy 
or Playtones.

From: 
asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 8:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] busy() not setting PRI_CAUSE

Okay, I think I need a sanity check here - If I call a person that's on the 
phone, I should get a busy signal.

Now more specifically, a call comes into the pbx via PRI.  The destination 
dialplan runs busy(20).  Now, the PRI causecode should get set to 17 (user 
busy) so that the originating end can play a busy tone, correct?

-Justin

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] dialplan =>how many concurrent calls

2014-07-10 Thread Rafael Visser 
Works fine..
Thanks Asghar!
rv


2014-07-10 9:35 GMT-04:00 Asghar Mohammad :

> you can use GROUP and GROUP_COUNT
>
> n,Set(GROUP()=aname)
> n,GotoIf($[${GROUP_COUNT(aname)} > 8]?${EXTEN},200)
> 200,Hangup
>
>
> On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser 
> wrote:
>
>> Hi guys.
>>
>> Does somebody knows how to get the concurrent calls from the dial plan?
>>
>> Or.
>>
>> How can i control not to run more than n simultaneus agi applications?
>>
>> Thanks in advance.
>> rv
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] dialplan =>how many concurrent calls

2014-07-10 Thread Asghar Mohammad 
you can use GROUP and GROUP_COUNT

n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} > 8]?${EXTEN},200)
200,Hangup


On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser 
wrote:

> Hi guys.
>
> Does somebody knows how to get the concurrent calls from the dial plan?
>
> Or.
>
> How can i control not to run more than n simultaneus agi applications?
>
> Thanks in advance.
> rv
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] dialplan =>how many concurrent calls

2014-07-10 Thread Rafael Visser 
Hi guys.

Does somebody knows how to get the concurrent calls from the dial plan?

Or.

How can i control not to run more than n simultaneus agi applications?

Thanks in advance.
rv
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digium E1 card stops working til disconnect machine power cord

2014-07-10 Thread A J Stiles 
On Thursday 10 Jul 2014, Ismael Gil wrote:
> Hi there,
> 
>  In one of my asterisk installation, there is a Digium E1 pri card
> connected. The asterisk and card are working properly.
>  The problem we have is that when a storm occurs in the area, the card
> stops working, and E1 lines connected not rise, even restart the machine.
> If however if you turn off the machine, disconnect the power cord,
> reattach it and turn on the machine again, E1 pri card correctly
> recognizes the E1 lines and the PBX starts to function properly. This only
> happens when there are storms in the area.
> 
>  Why is it necessary to disconnect the power cord for the card to work
> after a storm? How I can identify where the problem is?
>  How I can fix it?
> 
>  Thanks in advance,
> 
>  Ismaeleitor

It's a phenomenon called "latch-up", which occurs when extreme voltages are 
applied to the inputs of CMOS logic ICs.

A parasitic PNPN structure -- which is unavoidably present at each pin -- 
ordinarily provides some measure of static protection, by acting as diodes to 
clamp the voltage on any pin to the power rails; but it can also act like a 
PNP and NPN transistor in a sort of deadly embrace, where each one is holding 
the other in conduction.

If enough current flows into or out of an input pin under fault conditions, one 
of these transistors can turn on, which will cause the other transistor also 
to turn on; and the input can become clamped hard against one or other power 
rail.  This situation will persist as long as any current is flowing through 
the transistors, which in practice means until the supply is disconnected  
(or, in extremis, until the device overheats and melts down).


See also Wikipedia article: http://en.wikipedia.org/wiki/Latchup

-- 
AJS

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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue

2014-07-10 Thread Matthew Jordan 
On Thu, Jul 10, 2014 at 4:28 AM, Sameer Rathod  wrote:
> Hi Matt,
>
> I also tested the directmedia=yes over 3g connection ie with a public ip but
> I am getting only one way audio
> am I doing anything wrong?
>

If you are getting one way audio when direct media is enabled, then
one of the devices cannot find the other. This is most likely because
one of the devices is behind a NAT.

If both devices are truly publicly accessible, then you would have to
look at a pcap at the re-INVITEs sent to the devices to determine
where Asterisk told the devices to send their media, then debug your
network to determine why media could not be sent directly between
those two devices.

-- 
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

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Digium E1 card stops working til disconnect machine power cord

2014-07-10 Thread Ismael Gil 
Hi there,

In one of my asterisk installation, there is a Digium E1 pri card connected.
The asterisk and card are working properly.
The problem we have is that when a storm occurs in the area, the card stops working, and E1 lines connected not rise, even restart the machine.
If however if you turn off the machine, disconnect the power cord, reattach it and turn on the machine again, E1 pri card correctly recognizes the E1 lines and the PBX starts to function properly.
This only happens when there are storms in the area.

Why is it necessary to disconnect the power cord for the card to work after a storm?
How I can identify where the problem is?
How I can fix it?

Thanks in advance,

Ismaeleitor.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue

2014-07-10 Thread Sameer Rathod 
sorry I forgot to add the conf

sip.conf


[101]
type=friend
username=101
secret=101
host=dynamic
context=mkg
;nat=force_rport,comedia
;dtmfmode=rfc2833
;canreinvite=no
directmedia=yes
;directrtpsetup=yes
;avpf=yes
;encryption=yes
;disallow=all
;allow=ulaw
;icesupport=yes


[102]
type=friend
username=102
secret=101
host=dynamic
context=mkg
;nat=force_rport,comedia
;dtmfmode=rfc2833
;canreinvite=no
directmedia=yes
;directrtpsetup=yes
;avpf=yes
;encryption=yes
;disallow=all
;allow=ulaw
;icesupport=yes






On Thu, Jul 10, 2014 at 2:58 PM, Sameer Rathod  wrote:

> Hi Matt,
>
> I also tested the directmedia=yes over 3g connection ie with a public ip
> but I am getting only one way audio
> am I doing anything wrong?
>
>
> On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod 
> wrote:
>
>> Hi Matt,
>>
>> Thank you so much for explaining me this concept
>>
>> One more thing when I did testing for the above in different cases ie
>> with directmedia=yes and no I got the flow of packets attached with this
>> mail
>> Please have a look
>>
>> The flow stats that the rtp packet flows directly between end point
>> So as per above details probably it is due to both of my endpoints are on
>> the same network ie one side of the nat
>>
>> am i right?
>>
>>
>>
>>
>>
>>
>> On Wed, Jul 9, 2014 at 6:36 PM, Matthew Jordan 
>> wrote:
>>
>>> On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod 
>>> wrote:
>>> > Hi,
>>> >
>>> > with canreinvite=no and directmedia=no I and getting the message in
>>> the logs
>>> > for all calls
>>> >
>>> > "switching from simple_bridge technology to native_rtp"
>>> >
>>> >
>>> > -- Executing [102@mkg:1] Dial("SIP/101-0017", "SIP/102") in new
>>> stack
>>> >   == Using SIP RTP CoS mark 5
>>> > -- Called SIP/102
>>> > -- SIP/102-0018 is ringing
>>> > -- SIP/102-0018 answered SIP/101-0017
>>> > -- Channel SIP/101-0017 joined 'simple_bridge' basic-bridge
>>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>>> > -- Channel SIP/102-0018 joined 'simple_bridge' basic-bridge
>>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>>> >> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
>>> > simple_bridge technology to native_rtp
>>> >> 0x7f427c068a10 -- Probation passed - setting RTP source
>>> address to
>>> > 111.118.250.236:49344
>>> >> 0x7f427c068a10 -- Probation passed - setting RTP source
>>> address to
>>> > 111.118.250.236:49344
>>> >> 0x7f42500168d0 -- Probation passed - setting RTP source
>>> address to
>>> > 111.118.250.236:26326
>>> >> 0x7f42500168d0 -- Probation passed - setting RTP source
>>> address to
>>> > 111.118.250.236:26326
>>> > -- Channel SIP/101-0017 left 'native_rtp' basic-bridge
>>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>>> > -- Channel SIP/102-0018 left 'native_rtp' basic-bridge
>>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>>> >   == Spawn extension (mkg, 102, 1) exited non-zero on
>>> 'SIP/101-0017'
>>> >
>>> >
>>> >
>>> > I cannot understand why asterisk state diff bridges if all works same
>>> >
>>> > please can anyone explain me the working bridging concept and how to
>>> > configure and use bridges to route the rtp externally form asterisk.
>>> >
>>>
>>> I think I just answered this in your other thread, but I'll repeat it
>>> here.
>>>
>>> First, canreinvite has been deprecated as a naming convention for ...
>>> a long time. It's not even documented any more. The code will accept
>>> it, but all you're doing is setting the directmedia option twice:
>>>
>>> } else if (!strcasecmp(v->name, "directmedia") ||
>>> !strcasecmp(v->name, "canreinvite")) {
>>> ast_set_flag(&mask[0], SIP_REINVITE);
>>> ast_clear_flag(&flags[0], SIP_REINVITE);
>>>
>>> The native RTP bridge in Asterisk 12 manages bridges between two RTP
>>> capable channels. The bridge can either be formed remotely (in which
>>> case the media flows between the endpoints) or locally, in which case
>>> the media is swapped across the ports. It will attempt to perform a
>>> remote bridge if possible, while falling back to a local bridge if a
>>> remote bridge is not possible.
>>>
>>> In your particular case, you've explicitly told it to *not* do
>>> directmedia. So it won't perform a remote bridge.
>>>
>>> Even if you set directmedia=yes (or one of its variants), you may not
>>> have a successful remote bridge if one of the endpoints is behind a
>>> NAT. The sip.conf sample configuration documentation is actually quite
>>> good on this subject:
>>>
>>> ;--- MEDIA HANDLING
>>> 
>>> ; By default, Asterisk tries to re-invite media streams to an optimal
>>> path. If there's
>>> ; no reason for Asterisk to stay in the media path, the media will be
>>> redirected.
>>> ; This does not really work well in the case where Asterisk is outside
>>> and the
>>> ; clients are on the inside of a NAT. In that case, you want to set
>>> directmedia=nonat.
>>> ;
>>> ;directm

Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue

2014-07-10 Thread Sameer Rathod 
Hi Matt,

I also tested the directmedia=yes over 3g connection ie with a public ip
but I am getting only one way audio
am I doing anything wrong?


On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod  wrote:

> Hi Matt,
>
> Thank you so much for explaining me this concept
>
> One more thing when I did testing for the above in different cases ie with
> directmedia=yes and no I got the flow of packets attached with this mail
> Please have a look
>
> The flow stats that the rtp packet flows directly between end point
> So as per above details probably it is due to both of my endpoints are on
> the same network ie one side of the nat
>
> am i right?
>
>
>
>
>
>
> On Wed, Jul 9, 2014 at 6:36 PM, Matthew Jordan  wrote:
>
>> On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod 
>> wrote:
>> > Hi,
>> >
>> > with canreinvite=no and directmedia=no I and getting the message in the
>> logs
>> > for all calls
>> >
>> > "switching from simple_bridge technology to native_rtp"
>> >
>> >
>> > -- Executing [102@mkg:1] Dial("SIP/101-0017", "SIP/102") in new
>> stack
>> >   == Using SIP RTP CoS mark 5
>> > -- Called SIP/102
>> > -- SIP/102-0018 is ringing
>> > -- SIP/102-0018 answered SIP/101-0017
>> > -- Channel SIP/101-0017 joined 'simple_bridge' basic-bridge
>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>> > -- Channel SIP/102-0018 joined 'simple_bridge' basic-bridge
>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>> >> Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
>> > simple_bridge technology to native_rtp
>> >> 0x7f427c068a10 -- Probation passed - setting RTP source
>> address to
>> > 111.118.250.236:49344
>> >> 0x7f427c068a10 -- Probation passed - setting RTP source
>> address to
>> > 111.118.250.236:49344
>> >> 0x7f42500168d0 -- Probation passed - setting RTP source
>> address to
>> > 111.118.250.236:26326
>> >> 0x7f42500168d0 -- Probation passed - setting RTP source
>> address to
>> > 111.118.250.236:26326
>> > -- Channel SIP/101-0017 left 'native_rtp' basic-bridge
>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>> > -- Channel SIP/102-0018 left 'native_rtp' basic-bridge
>> > <0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab>
>> >   == Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-0017'
>> >
>> >
>> >
>> > I cannot understand why asterisk state diff bridges if all works same
>> >
>> > please can anyone explain me the working bridging concept and how to
>> > configure and use bridges to route the rtp externally form asterisk.
>> >
>>
>> I think I just answered this in your other thread, but I'll repeat it
>> here.
>>
>> First, canreinvite has been deprecated as a naming convention for ...
>> a long time. It's not even documented any more. The code will accept
>> it, but all you're doing is setting the directmedia option twice:
>>
>> } else if (!strcasecmp(v->name, "directmedia") ||
>> !strcasecmp(v->name, "canreinvite")) {
>> ast_set_flag(&mask[0], SIP_REINVITE);
>> ast_clear_flag(&flags[0], SIP_REINVITE);
>>
>> The native RTP bridge in Asterisk 12 manages bridges between two RTP
>> capable channels. The bridge can either be formed remotely (in which
>> case the media flows between the endpoints) or locally, in which case
>> the media is swapped across the ports. It will attempt to perform a
>> remote bridge if possible, while falling back to a local bridge if a
>> remote bridge is not possible.
>>
>> In your particular case, you've explicitly told it to *not* do
>> directmedia. So it won't perform a remote bridge.
>>
>> Even if you set directmedia=yes (or one of its variants), you may not
>> have a successful remote bridge if one of the endpoints is behind a
>> NAT. The sip.conf sample configuration documentation is actually quite
>> good on this subject:
>>
>> ;--- MEDIA HANDLING
>> 
>> ; By default, Asterisk tries to re-invite media streams to an optimal
>> path. If there's
>> ; no reason for Asterisk to stay in the media path, the media will be
>> redirected.
>> ; This does not really work well in the case where Asterisk is outside
>> and the
>> ; clients are on the inside of a NAT. In that case, you want to set
>> directmedia=nonat.
>> ;
>> ;directmedia=yes; Asterisk by default tries to redirect
>> the
>> ; RTP media stream to go directly from
>> ; the caller to the callee.  Some devices
>> do not
>> ; support this (especially if one of
>> them is behind a NAT).
>> ; The default setting is YES. If you
>> have all clients
>> ; behind a NAT, or for some other
>> reason want Asterisk to
>> ; stay in the audio path, you may want
>> to turn this off.
>>
>> ; This setting also affect direct RTP
>>