Re: [asterisk-users] Bug in Dial() string processing

[Antony Stone]

On Tuesday 27 October 2020 at 11:00:10, Antony Stone wrote:

> Hi.
> 
> I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1
> at least).

I've now confirmed that the same bug exists in 16.2.1

A Dial() command containing a SIP username/password combination which has a ! 
character in it is incorrectly parsed and fails.

> According to the documentation in channels/chan_sip.c the Dial() string
> syntax is:
> 
>  * SIP/devicename
>  * or SIP/username@domain (SIP uri)
>  * or SIP/username[:password[:md5secret[:authname[:transport@host[:port]
>  * or SIP/devicename/extension
>  * or SIP/devicename/extension/IPorHost
>  * or SIP/username@domain//IPorHost
>  *
>  * and there is an optional [!dnid] argument you can append to alter the
>  * To: header.
> 
> The bug is in the last part - the optional !dnid argument.
> 
> This should parse for a ! *following* any of the above syntaxes and assign
> whatever follows it to dnid.  If there is more than one ! in the entire
> string, dnid should follow the final one.
> 
> Unfortunately, the string parsing checks *first* for a ! and assigns
> everything following the first occurrence to dnid, and then attempts to
> parse whatever preceded it according to the above possibilities.
> 
> This means that if a SIP password, for example, contains a !, it is
> impossible to use the stated syntax
> 
> SIP/username[:password[:md5secret[:authname[:transport@host[:port]
> 
> Note that in this syntax, @host is non-optional, therefore parsing for a !
> preceding the @host part and regarding what follows the ! as dnid is
> incorrect.
> 
> 
> Regards,
> 
> 
> Antony.

-- 
"The future is already here.   It's just not evenly distributed yet."

 - William Gibson

   Please reply to the list;
 please *don't* CC me.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Bug in Dial() string processing

[Antony Stone]

Hi.

I've discovered a bug in the Dial() string processing (for Asterisk 13.14.1 at 
least).

According to the documentation in channels/chan_sip.c the Dial() string syntax 
is:

 * SIP/devicename
 * or SIP/username@domain (SIP uri)
 * or SIP/username[:password[:md5secret[:authname[:transport@host[:port]
 * or SIP/devicename/extension
 * or SIP/devicename/extension/IPorHost
 * or SIP/username@domain//IPorHost
 *
 * and there is an optional [!dnid] argument you can append to alter the
 * To: header.

The bug is in the last part - the optional !dnid argument.

This should parse for a ! *following* any of the above syntaxes and assign 
whatever follows it to dnid.  If there is more than one ! in the entire 
string, dnid should follow the final one.

Unfortunately, the string parsing checks *first* for a ! and assigns everything 
following the first occurrence to dnid, and then attempts to parse whatever 
preceded it according to the above possibilities.

This means that if a SIP password, for example, contains a !, it is impossible 
to use the stated syntax

SIP/username[:password[:md5secret[:authname[:transport@host[:port]

Note that in this syntax, @host is non-optional, therefore parsing for a ! 
preceding the @host part and regarding what follows the ! as dnid is 
incorrect.


Regards,


Antony.

-- 
+++ Divide By Cucumber Error.  Please Reinstall Universe And Reboot +++

   Please reply to the list;
 please *don't* CC me.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] bug in pjsip trust_id_outpound?

[Benoit Panizzon]

Hi Gang

If anyone else stumbles over the same Problem.

This is how I solved it for now:

On the IC Trunk:

trust_id_inbound=no => Makes sure the CallerID is taken from the From Header.
trust_id_outbound=yes => Does nothing useful, maybe a bug?
send_pai=no

On the incoming call, you have to pull the Privacy: header to figure
out the callerid presentation as asterisk is not setting this value if
you don't trust id inbound.

If the call is destined to a customer, you set CALLERPRES()=prohib
which makes sure, the From header is anonymized towards the customer.

If the call is destined to an IC, you have then to manually
PJSIP_HEADER(add) a P-Assertied-Identity and a Privacy Header correctly
setting the calling presentation. Asterisk is NOT doing this correctly.

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] bug in pjsip trust_id_outpound?

[Benoit Panizzon]

Hi Team

I'm still struggling to get privacy settings passed on correctly.

The Asterisk is sitting between customers and IC trunks.

On the customer face, of course I have those settings:

trust_id_inbound=yes
trust_id_outbound=no

This ensures that presentation is set to probibited, if the customer is
setting Privacy: ID.
It also ensures that the From: header is set to anonymous, hiding the
callerID if the caller requested presentations prohib.

So far, towards the customer side, this works as expected.

Towards the IC, we need to correctly populate the Request, From,
P-Asserted Identity and Privacy header. Sending From: anonymous is not
allowed.

So I set:

trust_id_inbound=no
trust_id_outbound=yes

Unfortunately I have to set inbound trust to no, to make sure the
Asterisk takes callerID from the From: header and NOT from the
P-Asserted Identity Header.

I then call pull the Privacy: Header with PJSIP_HEADER and set caller
presentation correctly.

But with outbound calls I am facing a HUGE problem.

I have set trust_id_outbound=yes. So I am expecting Asterisk to
correctly set the From: header to CallerID(number) and if
CallerID(num-pres) ist set to prohib, to add a Privacy: ID header.

This is not happening. From: is set to anonymous with missing Privacy:
header with the result, that the IC partner on the other side is
blocking this call.

Any idea how I could deal with this? Why is trust_id_outbound=yes not
behaving as expected?

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Bug in func_odbc module

[Bryant Zimmerman]

Hey all
  
 I have code we are moving from an early asterisk 13 system to the latest 
build. 
  
 The issue we are having is func_odbc calls are acting incorrectly. 
 We have tables that have fields with null values in them. 
  
 On the new system when we read a field with a null value it is copying the 
value from the previous filed into the value and not leaving the filed as 
null or blank string. So what is happening is we get variables inside of 
our dialplan that have values from other variables fields. 
 As soon as the system hits a value with a non null field, even a filed 
with an empty string it self corrects until it hits another field value 
with a null string in it. 
  
 Any thoughts on how and when this could get fixed?
  
 Thanks

Bryant 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

[Richard Mudgett]

On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner  wrote:

> I've had two Asterisk crashes today that seem to be caused by errors
> where chan->tech_pvt is pointing to something that can't be deallocated
> and I think I see a reference count bug in the above function.
>
> It contains:
>
> if (data->chan_old_vsrc) {
> ast_channel_unref(data->chan_old_vsrc);
> }
>
> Shouldn't this also have:
>
> data->chan_old_vsrc = NULL;
>
> It seems to me that if it doesn't and the next condition also isn't
> true, then the next time this same code is executed, it'll decrement
> the reference count of the old channel again, which is wrong since it
> hasn't been decremented.
>

Yes, doing that would be a good thing.  What you point out does leave a
dangling
channel pointer in data->chan_old_vsrc if the pointer is not set to NULL.
Please
create an issue for the dangling pointer.  The patch needs to be done for
v13+.

Richard
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] Bug in main/bridge.c:ast_bridge_update_talker_src_video_mode

[Richard Kenner]

I've had two Asterisk crashes today that seem to be caused by errors
where chan->tech_pvt is pointing to something that can't be deallocated
and I think I see a reference count bug in the above function.

It contains:

if (data->chan_old_vsrc) {
ast_channel_unref(data->chan_old_vsrc);
}

Shouldn't this also have:

data->chan_old_vsrc = NULL;

It seems to me that if it doesn't and the next condition also isn't
true, then the next time this same code is executed, it'll decrement
the reference count of the old channel again, which is wrong since it
hasn't been decremented.

What am I missing?

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] BUG or ???

[A J Stiles]

On Saturday 25 Feb 2017, Антон Сацкий wrote:
> Thanks U Richard
> i know about this solution
> but the main question why "${} substitution containing
> the SHELL is evaluated before anything else"

For the same reason why you do raising to powers before multiplications and 
divisions, and all those before you do additions and subtractions.  It's just 
a mathematical convention, agreed upon a long time ago and handed down 
verbatim from generation to generation ever since.

> Can U describe the rules   when and  why it happens?

The order goes something like this  (this is from memory; check docs for 
definitive version):

* ${} variable substitution / munging
* $[] expression evaluation
  - () round brackets
  - - arithmetical negation
  - * / % multiply and divide
  - + - add and subtract
  - < = > and =~ comparison
  - ! and ~ NOT
  - && and & AND
  - !! and | ^ OR and EOR
* ?: embedded IFELSE operator

Since ${} has a higher priority than ?: then the contents of the ${SHELL ...} 
expression will be evaluated *before* Asterisk decides whether to do the bit 
between the ? and the : or the bit after the : .  (And the most deepest 
embedded ${FROMEXTEN} needs to be evaluated before the shell can even be 
spawned to deal with the command).  Having performed the SHELL command and got 
a result for the ${}, it *then* begins dealing with the $[] expression. 
${CALLERID(num)} is compared against the target.  It's found to be false  
(empty string or numeric zero),  therefore the Set(var29=...) does *not* 
happen.  But, thanks to operator precedence, Asterisk has *already* evaluated 
what it would have set var29 to *before* it decides not to set it.

It is generally better to use a GotoIf() to skip past a step you don't want 
done, rather than embed it into an ExecIf() .  This is not the 1980s anymore, 
and the odd well-placed GOTO is perfectly OK  (and I'd reckon, actually *more* 
readable than the circumlocution necessary to avoid using it).

-- 
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 --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] BUG or ???

[Антон Сацкий]

Thanks U Richard
i know about this solution
but the main question why "${} substitution containing
the SHELL is evaluated before anything else"
Can U describe the rules   when and  why it happens?
Thanks

2017-02-24 23:44 GMT+02:00 Richard Mudgett :

>
>
> On Fri, Feb 24, 2017 at 3:30 PM, Антон Сацкий  wrote:
>
>> Got a strange situation
>>
>> [ext-queues]
>> ...
>> exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X
>> POST --header "Content-Type: application/json" --header "Accept:
>> application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "
>> http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}))
>>
>> exten => h,3,NoOp(${var29})
>> exten => h,4,Macro(hangupcall,)
>> ;--== end of [ext-queues] ==--;
>>
>>
>> so when i execute  it got
>>
>>
>> -- Executing [h@ext-queues:1] NoOp("SIP/100-0050", "100") in new
>> stack
>> -- Executing [h@ext-queues:2] ExecIf("SIP/100-0050", "0
>> ?Set(var29=[{"RequestedCount":0,"MissedCount":7,"Total":7}])") in new
>> stack
>> -- Executing [h@ext-queues:3] NoOp("SIP/100-0050", "") in new
>> stack
>> -- Executing [h@ext-queues:4] Macro("SIP/100-0050",
>> "hangupcall,") in new st
>>
>>
>> U can see  that Execif = 0 = falce   but   somehow
>> Shell ${SHELL(curl -X POST --header "Content-Type: application/json"
>> --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN},
>> \"Source\": \"asterisk\"}" "http://sIte.com:80/api/v1/cal
>> ls?apiKey=UABVAEI=3")}
>> executes and  get answer  from the server
>> [{"RequestedCount":0,"MissedCount":7,"Total":7}]
>>
>
> The Set isn't being executed by the ExecIf.  However the ${} substitution
> containing
> the SHELL is evaluated before anything else is examined.  This isn't a bug
> but the
> order of how things are evaluated.  You will have to do what you want
> another way:
>
> same = n,GotoIf($["${CALLERID(num)}" = ""]?skip)
> same = n,Set(var29=${SHELL(...)})
> same = n(skip),NoOp(${var29})
>
> Richard
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] BUG or ???

[Richard Mudgett]

On Fri, Feb 24, 2017 at 3:30 PM, Антон Сацкий  wrote:

> Got a strange situation
>
> [ext-queues]
> ...
> exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X
> POST --header "Content-Type: application/json" --header "Accept:
> application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "
> http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}))
>
> exten => h,3,NoOp(${var29})
> exten => h,4,Macro(hangupcall,)
> ;--== end of [ext-queues] ==--;
>
>
> so when i execute  it got
>
>
> -- Executing [h@ext-queues:1] NoOp("SIP/100-0050", "100") in new stack
> -- Executing [h@ext-queues:2] ExecIf("SIP/100-0050", "0
> ?Set(var29=[{"RequestedCount":0,"MissedCount":7,"Total":7}])") in new
> stack
> -- Executing [h@ext-queues:3] NoOp("SIP/100-0050", "") in new
> stack
> -- Executing [h@ext-queues:4] Macro("SIP/100-0050",
> "hangupcall,") in new st
>
>
> U can see  that Execif = 0 = falce   but   somehow
> Shell ${SHELL(curl -X POST --header "Content-Type: application/json"
> --header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN},
> \"Source\": \"asterisk\"}" "http://sIte.com:80/api/v1/
> calls?apiKey=UABVAEI=3")}
> executes and  get answer  from the server   [{"RequestedCount":0,"
> MissedCount":7,"Total":7}]
>

The Set isn't being executed by the ExecIf.  However the ${} substitution
containing
the SHELL is evaluated before anything else is examined.  This isn't a bug
but the
order of how things are evaluated.  You will have to do what you want
another way:

same = n,GotoIf($["${CALLERID(num)}" = ""]?skip)
same = n,Set(var29=${SHELL(...)})
same = n(skip),NoOp(${var29})

Richard
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

[asterisk-users] BUG or ???

[Антон Сацкий]

Got a strange situation

[ext-queues]
...
exten => h,2,ExecIf($[${CALLERID(num)} = ' ']?Set(var29=${SHELL(curl -X
POST --header "Content-Type: application/json" --header "Accept:
application/json" -d "{\"Phone\": ${FROMEXTEN}, \"Source\": \"asterisk\"}" "
http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}))

exten => h,3,NoOp(${var29})
exten => h,4,Macro(hangupcall,)
;--== end of [ext-queues] ==--;


so when i execute  it got


-- Executing [h@ext-queues:1] NoOp("SIP/100-0050", "100") in new stack
-- Executing [h@ext-queues:2] ExecIf("SIP/100-0050",
"0?Set(var29=[{"RequestedCount":0,"MissedCount":7,"Total":7}])")
in new stack
-- Executing [h@ext-queues:3] NoOp("SIP/100-0050", "") in new stack
-- Executing [h@ext-queues:4] Macro("SIP/100-0050", "hangupcall,")
in new st


U can see  that Execif = 0 = falce   but   somehow
Shell ${SHELL(curl -X POST --header "Content-Type: application/json"
--header "Accept: application/json" -d "{\"Phone\": ${FROMEXTEN},
\"Source\": \"asterisk\"}" "
http://sIte.com:80/api/v1/calls?apiKey=UABVAEI=3;)}
executes and  get answer  from the server
[{"RequestedCount":0,"MissedCount":7,"Total":7}]

i dont want it to be  executed



Thanks list for your help


-- 
Best regards
Antony
tel.   +380669197533
tel2. +380636564340
Paypal http://paypal.me/Satskiy

satski...@gmail.com 
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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

Re: [asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?

[Joshua Colp]

Richard Kenner wrote:

I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:

351 res = (int) *input * *value;

It's called from ast_frame_adjust_volume.

The frame looks like:

(gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer
= 100021, format = { id = AST_FORMAT_SLINEAR16, fattr = {format_attr
= { 0repeats 64 times}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64, src
= 0x51623b0 func_jitterbuffer interpolation, data = {ptr = 0x0,
uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}

so datalen is 0 and samples nonzero.  ast_frame_adjust_volume,
however, iterates over samples, not datalen.  Is that correct?

What does it mean to have a packet with a zero datalen anyway?


This is an interpolated frame from func_jitterbuffer. It's part of 
packet loss concealment. What scenario exposed this?


--
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

--
_
-- 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] Bug in ast_frame_adjust_volume in 12.2.0?

[Richard Kenner]

 This is an interpolated frame from func_jitterbuffer. It's part of 
 packet loss concealment. What scenario exposed this?

We were testing for clipping by doing Set(VOLUME(RX)=100) but we were
connecting to a ConfBridge that had a jitterbuffer.  This occurred when
the phone (SIP) hung up.

-- 
_
-- 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] Bug in ast_frame_adjust_volume in 12.2.0?

[Richard Kenner]

I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:

351 res = (int) *input * *value;

It's called from ast_frame_adjust_volume.

The frame looks like:

(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
  id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
  0 repeats 64 times}, rtp_marker_bit = 0 '\000'}}}, datalen = 0, 
  samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64, 
  src = 0x51623b0 func_jitterbuffer interpolation, data = {ptr = 0x0, 
uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0}, 
  flags = 0, ts = 0, len = 0, seqno = 0}

so datalen is 0 and samples nonzero.  ast_frame_adjust_volume, however,
iterates over samples, not datalen.  Is that correct?

What does it mean to have a packet with a zero datalen anyway?

-- 
_
-- 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] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call

[Justin Killen]

When doing a 'dahdi show channel X' from the asterisk console, when the line is 
not part of a call the echo cancellation line ALWAYS says 'currently OFF'.  
Once a call is in progress, it will change to 'ON'.  Is this a bug, or is the 
behavior by design?

My setup:

Asterisk 10.10.0
Dahdi 2.6.1
TE820 T1 card

In chan_dahdi.cfg:
...
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
...


And in #included chan_dahdi_additional.conf (this is a freepbx install):

...
;;[3884]
signalling=fxo_ks
pickupgroup=
mailbox=3884@default
immediate=no
echotraining=800
echocancelwhenbridged=yes
echocancel=yes
context=from-internal
callprogress=no
callgroup=
callerid=John Doe 3884
busydetect=no
busycount=7
accountcode=
channel=73
...

-Justin Killen
--
_
-- 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] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call

[Russ Meyerriecks]

On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote:
 When doing a 'dahdi show channel X' from the asterisk console, when the line
 is not part of a call the echo cancellation line ALWAYS says 'currently OFF'.
 Once a call is in progress, it will change to 'ON'.  Is this a bug, or is the
 behavior by design?

This is expected behavior. From a DAHDI perspective: The echocan chip's
channels are allocated on demand during channel setup and de-allocated on
channel teardown.

-- 
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com  www.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

Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo cancellation display is incorrect while not on a call

[Justin Killen]

This is highly confusing.  It would be nice if at least the display gave the 
configured value as well.

-Justin Killen

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks
Sent: Thursday, December 20, 2012 3:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug? 'dahdi show channel x' HWEC echo 
cancellation display is incorrect while not on a call

On Thu, Dec 20, 2012 at 03:13:01PM -0800, Justin Killen wrote:
 When doing a 'dahdi show channel X' from the asterisk console, when the line
 is not part of a call the echo cancellation line ALWAYS says 'currently OFF'.
 Once a call is in progress, it will change to 'ON'.  Is this a bug, or is the
 behavior by design?

This is expected behavior. From a DAHDI perspective: The echocan chip's
channels are allocated on demand during channel setup and de-allocated on
channel teardown.

-- 
Russ Meyerriecks
Digium, Inc. | Linux Kernel Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
direct: +1 256-428-6025
Check us out at: www.digium.com  www.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

--
_
-- 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] Bug or Not

[CDR]

I think I found another bug, but please let me know if there is a
workaround, since my bugs never get fixed.
In safe_asterisk, there is a section where the script executes some
startup scripts, located in /etc/asterisk/startup.d
However, when you restart asterisk with core restart now or you go
ahead and kill the asterisk process, these scripts that are so
important do not get executed.
The question is: where in the safe_asterisk script can I copy the
whole loop so in any event, if Asterisk gets restarted, these scripts
get properly executed. Otherwise there is no way to ensure that the
finite-state machine starts from an known start point.

--
_
-- 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] bug in queuemanager?

[Henry Dogger]

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Danny Nicholas]

Have you posted this to the forum Asterisk Support on asterisk.org?  One
thing I see is that you are doing an attended transfer (*2) vs a blind
transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension 204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension 203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps.

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200
that he should be at Queue 901 and transfers Customer 200 (using *2) to
Queue 901. Agent 301 now gets the call from Queue 901 with Customer 200,
answers the calls etc. After disconnect a new call arrivers immediately from
Queue 901, without any wrap-up time. This should be considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10' from
the asterisk CLI), it may shed some light on whether this is a bug or a
feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Henry Dogger]

Nope, I encounter this with blind transfer as well as attended
transfer

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?
One thing I see is that you are doing an attended transfer (*2) vs a
blind transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Danny Nicholas]

Do you have an isolated environment where you can do a core show channels
verbose after the transfer, but before the end of the call?  My suspicion
is that you are spawning a phantom local call.  Also, what does this
merriment look like in Master.csv?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Nope, I encounter this with blind transfer as well as attended transfer..

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?  One
thing I see is that you are doing an attended transfer (*2) vs a blind
transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension 204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension 203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps.

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200
that he should be at Queue 901 and transfers Customer 200 (using *2) to
Queue 901. Agent 301 now gets the call from Queue 901 with Customer 200,
answers the calls etc. After disconnect a new call arrivers immediately from
Queue 901, without any wrap-up time. This should be considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10' from
the asterisk CLI), it may shed some light on whether this is a bug or a
feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Henry Dogger]

Yes I do J here is the output http://pastebin.com/qpWqdA50

I don't put the cdr's in csv but in database, so not sure what you want
from the db J

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 16:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Do you have an isolated environment where you can do a core show
channels verbose after the transfer, but before the end of the call?
My suspicion is that you are spawning a phantom local call.  Also, what
does this merriment look like in Master.csv?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Nope, I encounter this with blind transfer as well as attended
transfer

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?
One thing I see is that you are doing an attended transfer (*2) vs a
blind transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Danny Nicholas]

Call 1 was from 346 to 900.   The log in the link provided shows it
correctly being in local/901 (line 8) from the queue and redialed (line 9).
Line 12 seems to be in sync with lines 8 and 9 but the one I would question
is line 11.  You have 4 active calls when it seems there should be 3 - 1
each for queue connect to agent and 1 for the transferred call.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Yes I do J here is the output http://pastebin.com/qpWqdA50

I don't put the cdr's in csv but in database, so not sure what you want from
the db J

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Do you have an isolated environment where you can do a core show channels
verbose after the transfer, but before the end of the call?  My suspicion
is that you are spawning a phantom local call.  Also, what does this
merriment look like in Master.csv?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Nope, I encounter this with blind transfer as well as attended transfer..

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?  One
thing I see is that you are doing an attended transfer (*2) vs a blind
transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension 204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension 203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps.

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer 200
that he should be at Queue 901 and transfers Customer 200 (using *2) to
Queue 901. Agent 301 now gets the call from Queue 901 with Customer 200,
answers the calls etc. After disconnect a new call arrivers immediately from
Queue 901, without any wrap-up time. This should be considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10' from
the asterisk CLI), it may shed some light on whether this is a bug or a
feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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

Re: [asterisk-users] bug in queuemanager?

[Henry Dogger]

Hm, I see what you mean.

What I must add, the phones are registered on a different PBX, and not
on the asterisk which is handling the queues...

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 17:37
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Call 1 was from 346 to 900.   The log in the link provided shows it
correctly being in local/901 (line 8) from the queue and redialed (line
9).  Line 12 seems to be in sync with lines 8 and 9 but the one I would
question is line 11.  You have 4 active calls when it seems there should
be 3 - 1 each for queue connect to agent and 1 for the transferred call.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Yes I do J here is the output http://pastebin.com/qpWqdA50

I don't put the cdr's in csv but in database, so not sure what you want
from the db J

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 16:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Do you have an isolated environment where you can do a core show
channels verbose after the transfer, but before the end of the call?
My suspicion is that you are spawning a phantom local call.  Also, what
does this merriment look like in Master.csv?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Nope, I encounter this with blind transfer as well as attended
transfer

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?
One thing I see is that you are doing an attended transfer (*2) vs a
blind transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation

Re: [asterisk-users] bug in queuemanager?

[Danny Nicholas]

The second PBX is just a Red Herring - once the call gets to the Asterisk
you did the core show channels from it's all really internal.  You might
be able to verify this if you have a local phone on that PBX to initiate the
900 call.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Hm, I see what you mean.

What I must add, the phones are registered on a different PBX, and not on
the asterisk which is handling the queues.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 17:37
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Call 1 was from 346 to 900.   The log in the link provided shows it
correctly being in local/901 (line 8) from the queue and redialed (line 9).
Line 12 seems to be in sync with lines 8 and 9 but the one I would question
is line 11.  You have 4 active calls when it seems there should be 3 - 1
each for queue connect to agent and 1 for the transferred call.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 10:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Yes I do J here is the output http://pastebin.com/qpWqdA50

I don't put the cdr's in csv but in database, so not sure what you want from
the db J

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:46
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Do you have an isolated environment where you can do a core show channels
verbose after the transfer, but before the end of the call?  My suspicion
is that you are spawning a phantom local call.  Also, what does this
merriment look like in Master.csv?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 9:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Nope, I encounter this with blind transfer as well as attended transfer..

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: maandag 7 november 2011 16:16
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] bug in queuemanager?

 

Have you posted this to the forum Asterisk Support on asterisk.org?  One
thing I see is that you are doing an attended transfer (*2) vs a blind
transfer (#1);  that could be causing some sort of problem.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: Monday, November 07, 2011 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Perhaps some help on where to look myself?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: donderdag 3 november 2011 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension 204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension 203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps.

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog

Re: [asterisk-users] bug in queuemanager?

[Henry Dogger]

Anyone?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry
Dogger
Sent: dinsdag 1 november 2011 13:00
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] bug in queuemanager?

[Henry Dogger]

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger h.dog...@telecats.nl
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a feature.  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

--
_
-- 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] Bug with Realtime?

[Dan Journo]

I checked the bug reports and all I could find was similar issues with the 
Asterisk 1.6 which (according to the reports) have been resolved.
I couldnt find anyone talking about 1.4 so I created a new issue and someone 
closed the case and added this note:-

 This does not appear to be a bug, but rather a support issue. Please use the 
 asterisk-users mailing list for such issues.
 The problem looks like your device has not re-registered after your 'sip 
 reload' which means it does not exist in memory, and thus causes Asterisk to 
 not know where to send the call. Your device needs to re-register after a 
 'sip reload' in order for Asterisk to know where to send the call.

I really think that sip reload shouldn't purge all the realtime peer 
registrations. It should treat the realtime peers the same way as the hardcoded 
peers. As i've said, the hardcoded peers don't lose registration when I issue a 
SIP RELOAD. 
Asterisk should be flexible enough to allow modification of the sip.conf file 
without losing all the realtime registrations.

Does anyone have a comment on the subject? Am I expecting too much?
I'm open to feedback.

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Carlos Chavez]

On Tue, 2010-09-21 at 19:04 -0400, Dan Journo wrote:
 I checked the bug reports and all I could find was similar issues with the 
 Asterisk 1.6 which (according to the reports) have been resolved.
 I couldnt find anyone talking about 1.4 so I created a new issue and someone 
 closed the case and added this note:-
 
  This does not appear to be a bug, but rather a support issue. Please use 
  the asterisk-users mailing list for such issues.
  The problem looks like your device has not re-registered after your 'sip 
  reload' which means it does not exist in memory, and thus causes Asterisk 
  to 
  not know where to send the call. Your device needs to re-register after a 
  'sip reload' in order for Asterisk to know where to send the call.
 
 I really think that sip reload shouldn't purge all the realtime peer 
 registrations. It should treat the realtime peers the same way as the 
 hardcoded peers. As i've said, the hardcoded peers don't lose registration 
 when I issue a SIP RELOAD. 
 Asterisk should be flexible enough to allow modification of the sip.conf file 
 without losing all the realtime registrations.
 
 Does anyone have a comment on the subject? Am I expecting too much?
 I'm open to feedback.
 
I use realtime on 1.4 and 1.6 servers but always with
rtcachefriends=yes in sip.conf so I can use things like sip show peers.
My experience is that when I issue a sip reload all registrations
disappear but the moment a call comes in for a peer it uses the last IP
where it was registered to send the call.  I guess this is the
equivalent do doing sip show peer XXX load.  The peer does not need to
register again to get calls.

I think this works because the internal database has the last IP of the
peer even after a sip reload.  If you do a database show you will see
something like:

/SIP/Registry/
192.168.2.215:5060:3600::sip:x...@192.168.2.215:5060;transport=udp

-- 
Carlos Chavez
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 I use realtime on 1.4 and 1.6 servers but always with rtcachefriends=yes in 
 sip.conf

I already use that and it doesnt seem to re-register when a call comes in. 
Only when the registration period expires, or the peer dials out.
-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 Check the SIP debug and see what is going on. 
 Leif.

Hi,

I checked the SIP debug.

As soon as I issue the RELOAD command, no SIP data gets transferred to the 
phone.

Asterisk output: http://pastebin.com/FB675N16

Any ideas how I can do a SIP reload without losing the Sip Phones registration?

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Peder]

I am not aware of any way to do that.  My question is if you are using
realtime, why are you doing a sip reload?  If you change the settings on a
device in the realtime DB, just prune it and it will grab the new config the
next time they re-register.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Monday, September 20, 2010 10:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

 Check the SIP debug and see what is going on. 
 Leif.

Hi,

I checked the SIP debug.

As soon as I issue the RELOAD command, no SIP data gets transferred to the
phone.

Asterisk output: http://pastebin.com/FB675N16

Any ideas how I can do a SIP reload without losing the Sip Phones
registration?

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

Can we not do pastebin any more?

I just received this:-

[PASTEBIN URL REMOVED] has been detected as suspicious URLs,and Quarantine to 
user's spam folder has been taken on 9/20/2010 8:24:38 AM.
Message details:
Server: MADRID
Sender: d...@keshrcommunications.com;
Recipient: asterisk-users@lists.digium.com;
Subject: Suspicious URL:Re: [asterisk-users] Bug with Realtime?

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 My question is if you are using realtime, why are you doing a sip reload?

I said previously:-

 Let's say I add a new provider to my service and therefore have to add 
 another register= command into sip.conf, I'd have to issue a sip reload 
 which would kill off all the realtime sip phones.

Unless I can do register= in realtime too?

Dan

-- 
_
-- 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] Bug with Realtime?

[Roger Burton West]

On Mon, Sep 20, 2010 at 11:59:16AM -0400, Dan Journo wrote:
Can we not do pastebin any more?

No, it's just one user with an excessively paranoid and chatty
mailfilter.


-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 Check the SIP debug and see what is going on. Alternatively you could turn 
 off 
the qualify option with qualify=no.

I'll take a look at the sip debug, but qualify needs to stay on, so thats not 
an option.


-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 That's not a bug. Only when the phone registers or performs some sort of 
 action 
 (such as placing a call, etc...) does Asterisk query the database. If your 
 phones have a short re-registration time this becomes less of a problem.

How do you explain that as soon as I issue a reload command, the realtime 
phones stop receiving calls?
To test your theory, I rebooted the phone so that it had a fresh registration, 
I made and receives calls successfully, then issued a 'reload', then trying to 
dial in again, and the phone didnt ring.

After a few seconds, the CLI says:-
[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer: Peer 
'kesher_201' is now UNREACHABLE!  Last qualify: 58

How can this not be a bug? The phone works fine for hours, and then as soon as 
I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the reload. It just stops receiving 
calls after a reload.

I want to move my clients over to realtime so they can manage their accounts 
online, but I cant do that if they become UNREACHABLE when I do a reload.

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Peder]

A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.  If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine.  Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime.  I
agree that it seems wrong and even argued that several years ago when this
feature came out, but it is what it is.  As someone else said, the answer is
don't do a 'reload', do an extensions reload or whatever it is specific
to your changes.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

 That's not a bug. Only when the phone registers or performs some sort of
action 
 (such as placing a call, etc...) does Asterisk query the database. If your

 phones have a short re-registration time this becomes less of a problem.

How do you explain that as soon as I issue a reload command, the realtime
phones stop receiving calls?
To test your theory, I rebooted the phone so that it had a fresh
registration, I made and receives calls successfully, then issued a
'reload', then trying to dial in again, and the phone didnt ring.

After a few seconds, the CLI says:-
[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer:
Peer 'kesher_201' is now UNREACHABLE!  Last qualify: 58

How can this not be a bug? The phone works fine for hours, and then as soon
as I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the reload. It just stops
receiving calls after a reload.

I want to move my clients over to realtime so they can manage their accounts
online, but I cant do that if they become UNREACHABLE when I do a reload.

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

 That's not a bug. Only when the phone registers or performs some sort of
action 
 (such as placing a call, etc...) does Asterisk query the database. If
your 
 phones have a short re-registration time this becomes less of a problem.

How do you explain that as soon as I issue a reload command, the
realtime phones stop receiving calls?
To test your theory, I rebooted the phone so that it had a fresh
registration, I made and receives calls successfully, then issued a
'reload', then trying to dial in again, and the phone didnt ring.

After a few seconds, the CLI says:-
[2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer:
Peer 'kesher_201' is now UNREACHABLE!  Last qualify: 58

How can this not be a bug? The phone works fine for hours, and then as soon
as I issue a reload command, its UNREACHABLE.
ps. The phone can still make calls after the reload. It just stops
receiving calls after a reload.

I want to move my clients over to realtime so they can manage their
accounts online, but I cant do that if they become UNREACHABLE when I do a
reload.

Thanks
Dan

I'm still going to defend that this is not a bug.  It is up to you to
insure that your phones are in constant connectivity without burdening your
bandwith.  You could be aggressive and register your phone every 30 seconds;
a more realistic approach would be a register every 90-120 seconds (I have
Polycom 501's and they take at least 90 seconds to come back up on a
restart).  If your clients can't take 2 minutes of downtime on a phone,
they don't need to be on VOIP.


-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone. 

Finally an answer that seemed more realistic. But it doesnt explain why the 
phones that are hard coded in the sip.conf file don't lose registration.

Any ideas?

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 As someone else said, the answer is
don't do a 'reload', do an extensions reload or whatever it is specific
to your changes.

You are correct. I'm just being lazy. But I'm just worried that some time in 
the future, I'll have to reload the sip config, and therefore flush out all the 
realtime phones.

Can asterisk cope with 1000 phones all re-registering every 60 seconds?

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[John Novack]

Danny Nicholas wrote:
snip
If your clients can't take 2 minutes of downtime on a phone, they 
don't need to be on VOIP.




If VOIP ( and Asterisk ) ever really expect to be the future of 
Telephony   this ( attitude ) has to change

90 percent availability is unacceptable, even 95 percent,  for that 
matter even 99.99 percent availability.
Nothing short of 99.999 percent available will win.

Just one old phone man's opinion. ( along with a large number of users )

John Novack

-- 

Dog is my Co-pilot


-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 9:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

 As someone else said, the answer is
don't do a 'reload', do an extensions reload or whatever it is specific
to your changes.

You are correct. I'm just being lazy. But I'm just worried that some time in
the future, I'll have to reload the sip config, and therefore flush out all
the realtime phones.

Can asterisk cope with 1000 phones all re-registering every 60 seconds?

Thanks
Dan

Asterisk absolutely can; The more valid questions are these:
Is my CPU/Bandwidth stout enough for 1000 extra registrations per
minute?
Should I make a daemon instead to check and register UNKNOWN peers?




-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Thursday, September 16, 2010 9:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

Danny Nicholas wrote:
snip
If your clients can't take 2 minutes of downtime on a phone, they 
don't need to be on VOIP.


If VOIP ( and Asterisk ) ever really expect to be the future of 
Telephony   this ( attitude ) has to change

90 percent availability is unacceptable, even 95 percent,  for that 
matter even 99.99 percent availability.
Nothing short of 99.999 percent available will win.

Just one old phone man's opinion. ( along with a large number of users )

John Novack

Noted - but if OP does a reload once a day, 120 seconds (2 minutes) out of
1 day (14400 seconds) is 99.17% uptime; close enough to 99.999 percent in
most folks books.  What percentage of businesses use their phones 24/7?


-- 
_
-- 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] Bug with Realtime?

[Zeeshan Zakaria]

When making an outbound call, if sip peer is not registered, first it
registers itself, and then makes the call. This is why you don't see any
problem dialing out. For receiving, asterisk has to wait until the sip peer
registers, otherwise asterisk has nowhere to send the call.

I know the pain, as I deal with the same situation. So I don't do 'reload'
or 'sip reload' except if sip password (secret) has been changed, in which
case I prefer to use 'sip prune realtime peer extension' followed by 'sip
show peer extension load'. Most of the sip devices re-register every 60
seconds, or if they don't on a realtime network, depending upon the
bandwidth, they should be made to do so. Or in some cases you can send a
reboot signal to a sip device too. The bottom line is, try not to do a
'reload' as it would affect everybody else too by dropping their
registrations temporarily.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-16 10:04 AM, Peder pe...@networkoblivion.com wrote:

A reload flushes the SIP registration database, so once you do a reload,
that phones reg is gone.  If the reg is set for a short period, say 60
seconds, then in 60 seconds it will re-register and work fine.  Yes, it is a
total pain, but this is the way it has worked since day 1 for realtime.  I
agree that it seems wrong and even argued that several years ago when this
feature came out, but it is what it is.  As someone else said, the answer is
don't do a 'reload', do an extensions reload or whatever it is specific
to your changes.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-bo...

Sent: Thursday, September 16, 2010 8:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussi...

Subject: Re: [asterisk-users] Bug with Realtime?

 That's not a bug. Only when the phone registers ...
-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 Noted - but if OP does a reload once a day, 120 seconds (2 minutes) out of
 1 day (14400 seconds) is 99.17% uptime; close enough to 99.999 percent in
 most folks books.  What percentage of businesses use their phones 24/7?

Even if its once a month, it's still too much in my book. No wonder people have 
had bad experiences with voip.
If I'm going to run my service on a platform that will inevitably have some 
downtime, I wouldn't be a very good business man!

Let's say I add a new provider to my service and therefore have to add another 
register= command into sip.conf, I'd have to issue a sip reload which 
would kill off all the realtime sip phones.

 A reload flushes the SIP registration database, so once you do a 
 reload, that phones reg is gone.

Surely you could move the flush part of sip reload into a separate command 
and therefore ensure the realtime phones stay connected properly.
Still doesnt explain why, when I do a reload, only the realtime users are 
affected, and not the hard coded ones.

Dan

-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 10:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

 Noted - but if OP does a reload once a day, 120 seconds (2 minutes) out
of
 1 day (14400 seconds) is 99.17% uptime; close enough to 99.999 percent
in
 most folks books.  What percentage of businesses use their phones 24/7?

Even if its once a month, it's still too much in my book. No wonder people
have had bad experiences with voip.
If I'm going to run my service on a platform that will inevitably have some
downtime, I wouldn't be a very good business man!

Let's say I add a new provider to my service and therefore have to add
another register= command into sip.conf, I'd have to issue a sip reload
which would kill off all the realtime sip phones.

 A reload flushes the SIP registration database, so once you do a 
 reload, that phones reg is gone.

Surely you could move the flush part of sip reload into a separate
command and therefore ensure the realtime phones stay connected properly.
Still doesnt explain why, when I do a reload, only the realtime users are
affected, and not the hard coded ones.

Dan

Just a WAG, but I'm betting that the hard-coded users are loaded into a hash
that Asterisk handles and the real-time users are either in a different hash
or not at all.  Since real-time is an add-on vs a built-in feature, this
might indeed be a bug or shortcoming.  Have you checked the Issue Tracker
for this?


-- 
_
-- 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] Bug with Realtime?

[Peder]

 But it doesnt explain why the phones that are hard coded in the sip.conf
file don't lose registration.

On a reload, it re-reads the sip.conf config file and sees the users in
there, so it doesn't flush them.  It doesn't pull down the whole SIP table
on a reload, it only loads a realtime peer config when the phone tries to
register, so realtime users disappear on a reload, but sip.conf users do
not.

PA



-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

  Have you checked the Issue Tracker

Not yet. I wanted to see if it's just me before searching through/raising a bug 
report.

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

Is there any development work being done on the realtime addon? Theres been no 
updates since April.

-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Thursday, September 16, 2010 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bug with Realtime?

  Have you checked the Issue Tracker

Not yet. I wanted to see if it's just me before searching through/raising a
bug report.

Always a good idea to check it anyway;  somebody else may have put the issue
there without writing on the list.


-- 
_
-- 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] Bug with Realtime?

[Tilghman Lesher]

On Thursday 16 September 2010 11:23:37 Dan Journo wrote:
 Is there any development work being done on the realtime addon? Theres been
 no updates since April.

Realtime is integrated into the core; it is not an addon.  Perhaps you're
referring to the mysql realtime driver?  The driver modules tend not to need
updates that often; but only when the API changes.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.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

Re: [asterisk-users] Bug with Realtime?

[Leif Madsen]

On 10-09-16 09:43 AM, Dan Journo wrote:
 That's not a bug. Only when the phone registers or performs some sort of 
 action
 (such as placing a call, etc...) does Asterisk query the database. If your
 phones have a short re-registration time this becomes less of a problem.

 How do you explain that as soon as I issue a reload command, the realtime 
 phones stop receiving calls?
 To test your theory, I rebooted the phone so that it had a fresh 
 registration, I made and receives calls successfully, then issued a 'reload', 
 then trying to dial in again, and the phone didnt ring.

 After a few seconds, the CLI says:-
 [2010-09-16 14:39:29] NOTICE[24611]: chan_sip.c:17200 sip_poke_noanswer: Peer 
 'kesher_201' is now UNREACHABLE!  Last qualify: 58

 How can this not be a bug? The phone works fine for hours, and then as soon 
 as I issue a reload command, its UNREACHABLE.
 ps. The phone can still make calls after the reload. It just stops 
 receiving calls after a reload.

That sounds like a qualify issue in that the phone does not respond to a NOTIFY 
message.

Check the SIP debug and see what is going on. Alternatively you could turn off 
the qualify option with qualify=no.

Leif.

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

Hi,

I think ive found a bug but need someone to double check.

Whenever I issue a reload in Asterisk, any realtime extensions stop receiving 
calls.

I have to reboot the sip phones in order to get them to re-register.

Can anyone see if they have a similar problem?

Asterisk 1.4.32
Mysql realtime.

Thanks
Dan

-- 
_
-- 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] Bug with Realtime?

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, September 15, 2010 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Bug with Realtime?

 

Hi,


I think ive found a bug but need someone to double check.

 

Whenever I issue a reload in Asterisk, any realtime extensions stop
receiving calls.

 

I have to reboot the sip phones in order to get them to re-register.

 

Can anyone see if they have a similar problem?

 

Asterisk 1.4.32

Mysql realtime.

 

Thanks

Dan

 

By reload you mean sip reload or just any reload in general?

-- 
_
-- 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] Bug with Realtime?

[Dan Journo]

 By reload you mean sip reload or just any reload in general?

Reload in general.

It might be an issue only with the Polycom sip phones. Not been able to test 
any others. I'll try a software phone tomorrow.
-- 
_
-- 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] Bug with Realtime?

[Jonas Kellens]

On 09/15/2010 09:41 PM, Dan Journo wrote:


Hi,


I think ive found a bug but need someone to double check.

Whenever I issue a reload in Asterisk, any realtime extensions stop 
receiving calls.


I have to reboot the sip phones in order to get them to re-register.

Can anyone see if they have a similar problem?

Asterisk 1.4.32

Mysql realtime.

Thanks

Dan



Yes you loose all SIP registrations and they need to re-register to be 
reachable again. Don't know if this is a bug, but it's like that in 1.4 
and 1.6.2.



Jonas.
-- 
_
-- 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] Bug with Realtime?

[Leif Madsen]

On 10-09-15 03:41 PM, Dan Journo wrote:
 I think ive found a bug but need someone to double check.

 Whenever I issue a reload in Asterisk, any realtime extensions stop
 receiving calls.

 I have to reboot the sip phones in order to get them to re-register.

 Can anyone see if they have a similar problem?

 Asterisk 1.4.32

 Mysql realtime.

That's not a bug. Only when the phone registers or performs some sort of action 
(such as placing a call, etc...) does Asterisk query the database. If your 
phones have a short re-registration time this becomes less of a problem.

Leif.

-- 
_
-- 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] Bug with Realtime?

[Zeeshan Zakaria]

You can do 'extensions reload' or 'ael reload' if you don't want to lose
real-time sip registrations. I only reload what is needed to be reloaded
instead of reloading everything.

Zeeshan A Zakaria

--
www.ilovetovoip.com

On 2010-09-15 4:28 PM, Leif Madsen leif.mad...@asteriskdocs.org wrote:

On 10-09-15 03:41 PM, Dan Journo wrote:
 I think ive found a bug but need someone to double check.
...
That's not a bug. Only when the phone registers or performs some sort of
action
(such as placing a call, etc...) does Asterisk query the database. If your
phones have a short re-registration time this becomes less of a problem.

Leif.

--
_
-- 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] bug with Moh on MeetMe ?

[Mickael Monsieur]

Hello,
The MeetMe application refuses MusicOnHold personalized and skip always in
the default!
Have you any idea how to fix this?

-- Executing [028883...@default:1] Set(SIP/109.10.214.1-0002,
CHANNEL(language)=fr) in new stack
-- Executing [028883...@default:2] Answer(SIP/109.10.214.1-0002,
) in new stack
-- Executing [028883...@default:3] Playback(SIP/109.10.214.1-0002,
welcome) in new stack
-- SIP/109.10.214.1-0002 Playing 'welcome.alaw' (language 'fr')
[Jun 13 12:30:00] NOTICE[13437]: channel.c:3012 __ast_read: Dropping
incompatible voice frame on SIP/109.10.214.1-0002 of format ulaw since
our native format has changed to 0x8 (alaw)
-- Executing [028883...@default:4]
MeetMeCount(SIP/109.10.214.1-0002, 100,COUNT) in new stack
  == Parsing '/etc/asterisk/meetme.conf':   == Found
-- Executing [028883...@default:5] GotoIf(SIP/109.10.214.1-0002,
0?100) in new stack
-- Executing [028883...@default:6] MeetMe(SIP/109.10.214.1-0002,
100,1pdM(*personnalised*)) in new stack
-- Created MeetMe conference 1023 for conference '100'
-- *Started music on hold*, class *'personnalised*', on
SIP/109.10.214.1-0002
-- *Stopped music on hold* on SIP/109.10.214.1-0002
-- *Started music on hold, class 'default',* on
SIP/109.10.214.1-0002

Thank you,
Mickael.
-- 
_
-- 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] bug with Moh on MeetMe ?

[Daniel Knoll]

Hi Michael,
Can you show us the output from:
moh show classes and moh show files Command

Or try it to set a new exten after setting the language with:
exten = 12345,n,Set(CHANNEL(musicclass)=personalised)

Daniel


Am 13.06.2010 um 12:35 schrieb Mickael Monsieur:

 Hello,
 The MeetMe application refuses MusicOnHold personalized and skip always in 
 the default!
 Have you any idea how to fix this?
 
 -- Executing [028883...@default:1] Set(SIP/109.10.214.1-0002, 
 CHANNEL(language)=fr) in new stack
 -- Executing [028883...@default:2] Answer(SIP/109.10.214.1-0002, 
 ) in new stack
 -- Executing [028883...@default:3] Playback(SIP/109.10.214.1-0002, 
 welcome) in new stack
 -- SIP/109.10.214.1-0002 Playing 'welcome.alaw' (language 'fr')
 [Jun 13 12:30:00] NOTICE[13437]: channel.c:3012 __ast_read: Dropping 
 incompatible voice frame on SIP/109.10.214.1-0002 of format ulaw since 
 our native format has changed to 0x8 (alaw)
 -- Executing [028883...@default:4] 
 MeetMeCount(SIP/109.10.214.1-0002, 100,COUNT) in new stack
   == Parsing '/etc/asterisk/meetme.conf':   == Found
 -- Executing [028883...@default:5] GotoIf(SIP/109.10.214.1-0002, 
 0?100) in new stack
 -- Executing [028883...@default:6] MeetMe(SIP/109.10.214.1-0002, 
 100,1pdM(personnalised)) in new stack
 -- Created MeetMe conference 1023 for conference '100'
 -- Started music on hold, class 'personnalised', on 
 SIP/109.10.214.1-0002
 -- Stopped music on hold on SIP/109.10.214.1-0002
 -- Started music on hold, class 'default', on SIP/109.10.214.1-0002
 
 Thank you,
 Mickael.
 -- 
 _
 -- 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

Daniel Knoll

Liberdastr. 9 
12047 Berlin

fon +49 (0)179 20 16 50 8
mail dan...@danielknoll.de
web www.danielknoll.de





-- 
_
-- 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] bug with Moh on MeetMe ?

[Philipp von Klitzing]

Look!

 refuses MusicOnHold personalized
  -- Started music on hold, class 'personnalised'

Can you see it?! Two typos. ;-

Philipp


-- 
_
-- 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] bug in asterisk

[benoit bernard]

Hello all,

i have asterisk  installed  in our call centre and we work 24h in day with
this server ,the problem is each day in the night the server hangs and the
calls stopped

And i must to restart asterisk with this command “service asterisk restart”

When i make service asterisk start  i got the message failed that is mean
that the service is already ON

Any help please
-- 
_
-- 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] bug in asterisk

[Mindaugas Kezys]

Execute such commands with cronjob every night:

/etc/init.d/asterisk stop
sleep 3
killall -9 safe_asterisk
killall -9 asterisk
/etc/init.d/asterisk start

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of benoit bernard
Sent: Tuesday, May 11, 2010 1:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] bug in asterisk

Hello all,
 
i have asterisk  installed  in our call centre and we work 24h in day with
this server ,the problem is each day in the night the server hangs and the
calls stopped 
 
And i must to restart asterisk with this command “service asterisk restart”
 
When i make service asterisk start  i got the message failed that is mean
that the service is already ON
 
Any help please
 


-- 
_
-- 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] bug in asterisk

[Vardan]

Hello
I think this is not right way :)
Look the log's files, find the problem and resolv
The cronjob is the way to stay fat always online, until you find the 
problem :)

Vardan

Mindaugas Kezys wrote:
 Execute such commands with cronjob every night:

 /etc/init.d/asterisk stop
 sleep 3
 killall -9 safe_asterisk
 killall -9 asterisk
 /etc/init.d/asterisk start

 Regards,
 Mindaugas Kezys

 Kolmisoft UAB
 VoIP Billing Solutions
 e-mail: i...@kolmisoft.com
 URL: http://www.kolmisoft.com


 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of benoit bernard
 Sent: Tuesday, May 11, 2010 1:51 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] bug in asterisk

 Hello all,

 i have asterisk  installed  in our call centre and we work 24h in day with
 this server ,the problem is each day in the night the server hangs and the
 calls stopped

 And i must to restart asterisk with this command “service asterisk restart”

 When i make service asterisk start  i got the message failed that is mean
 that the service is already ON

 Any help please





-- 
_
-- 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] bug in asterisk

[Mindaugas Kezys]

Check Asterisk changelog
(http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.31) with
ctrl+f deadlock.

Guess how many deadlock related bugs wonderful Digium programmers will solve
in future releases?

My proposition is not solution to the problem, its the survival guide in
Asterisk world.

We have 470 servers deployed around the world with Asterisk and this piece
of code extended my and mine coworkers lifes by many years. 

If you really want to solve your problem - start here:
http://www.voip-info.org/wiki/view/Asterisk+debugging 

Regards,
Mindaugas Kezys

Kolmisoft UAB 
VoIP Billing Solutions
e-mail: i...@kolmisoft.com
URL: http://www.kolmisoft.com


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vardan
Sent: Tuesday, May 11, 2010 2:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] bug in asterisk

Hello
I think this is not right way :)
Look the log's files, find the problem and resolv
The cronjob is the way to stay fat always online, until you find the 
problem :)

Vardan

Mindaugas Kezys wrote:
 Execute such commands with cronjob every night:

 /etc/init.d/asterisk stop
 sleep 3
 killall -9 safe_asterisk
 killall -9 asterisk
 /etc/init.d/asterisk start

 Regards,
 Mindaugas Kezys

 Kolmisoft UAB
 VoIP Billing Solutions
 e-mail: i...@kolmisoft.com
 URL: http://www.kolmisoft.com


 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of benoit
bernard
 Sent: Tuesday, May 11, 2010 1:51 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] bug in asterisk

 Hello all,

 i have asterisk  installed  in our call centre and we work 24h in day with
 this server ,the problem is each day in the night the server hangs and the
 calls stopped

 And i must to restart asterisk with this command service asterisk
restart

 When i make service asterisk start  i got the message failed that is mean
 that the service is already ON

 Any help please





-- 
_
-- 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] Bug or feature: comments in chan_dahdi.conf.sample

[Olivier]

Hi,

1. From chan_dahdi.conf.sample (asterisk 1.6.1.18) :
; Switchtype:  Only used for PRI.
;
; national:   National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess:   ATT 4ESS
; 5ess:   Lucent 5ESS
; euroisdn:   EuroISDN (common in Europe)
; ni1:Old National ISDN 1
; qsig:   Q.SIG
;
;switchtype=euroisdn

At the same time, dahdi_genconf generates files in which there is:
; Span 1: B4/0/1 B4XXP (PCI) Card 0 Span 1 (MASTER) AMI/CCS
group=1,11
context=remote
switchtype = euroisdn
signalling = bri_cpe
channel = 1-2
context = default
group = 63


Does switchtype parameter also apply to BRI ?
If positive, should we change this ; Switchtype:  Only used for PRI
comment ?

2. In chan_dahdi.conf.sample, you can also read ; national:   National
ISDN 2 (default).
Using dahdi_genconf and without changing any default, I can see that
switchtype  is defaulted to euroisdn.
Should this Switchtype default comment be changed ?

Regards
-- 
_
-- 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] Bug or feature: cdr_odbc.conf.sample

[Olivier]

Hello,

From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample :
;
; cdr_odbc.conf
;

;[global]
;dsn=MySQL-test
;username=username
;password=password
;loguniqueid=yes
;dispositionstring=yes
;table=cdr  ;cdr is default table name
;usegmtime=no ; set to yes to log in GMT


Though, reading from https://issues.asterisk.org/view.php?id=15021, it seems
that lines username= and password= in cdr_odbc.conf are not used anymore
(the fields in res_odbc.conf are used instead).

My question are :
1. Are those lines still used in 1.6.1.X ?
2. If those lines are not used anymore, would you think more appropriate to
remove them from cdr_odbc.conf.sample ?

Regards
-- 
_
-- 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] Bug or feature: cdr_odbc.conf.sample

[Tilghman Lesher]

On Sunday 18 April 2010 04:10:11 Olivier wrote:
 From asterisk-1.6.1.18/configs/cdr_odbc.conf.sample :

 ;
 ; cdr_odbc.conf
 ;

 ;[global]
 ;dsn=MySQL-test
 ;username=username
 ;password=password
 ;loguniqueid=yes
 ;dispositionstring=yes
 ;table=cdr  ;cdr is default table name
 ;usegmtime=no ; set to yes to log in GMT

 Though, reading from https://issues.asterisk.org/view.php?id=15021, it
 seems that lines username= and password= in cdr_odbc.conf are not used
 anymore (the fields in res_odbc.conf are used instead).

 My question are :
 1. Are those lines still used in 1.6.1.X ?

No.

 2. If those lines are not used anymore, would you think more appropriate to
 remove them from cdr_odbc.conf.sample ?

Removed, as of revision 257770.  Thank you for the reminder.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.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

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

[Faidon Liambotis]

Justin Piszcz wrote:
  Found root cause-- root cause is asterisk PBX software.  I use an
SPA3102.
 When someone called me, they accidentally dropped the connection, I called
 them back in a short period.  It is during this time (and the last time)
 this happened that the box froze under multiple(!) kernels, always when
 someone was calling.
snip
 I don't know what asterisk is doing but top did run before the crash
 and asterisk was using 100% CPU and as I noted before all other processes
 were in D-state.
 
 When this bug occurs, it freezes I/O to all devices and the only way to
 recover
 is to reboot the system.
That's obviously *not* the root cause.

It's not normal for an application that isn't even privileged to hang
all I/O and, subsequently everything on a system.

This is almost probably a kernel issue and asterisk just does something
that triggers this bug.

Regards,
Faidon

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

[Justin Piszcz]

On Sat, 21 Nov 2009, Faidon Liambotis wrote:

 Justin Piszcz wrote:
  Found root cause-- root cause is asterisk PBX software.  I use an
 SPA3102.
 When someone called me, they accidentally dropped the connection, I called
 them back in a short period.  It is during this time (and the last time)
 this happened that the box froze under multiple(!) kernels, always when
 someone was calling.
 snip
 I don't know what asterisk is doing but top did run before the crash
 and asterisk was using 100% CPU and as I noted before all other processes
 were in D-state.

 When this bug occurs, it freezes I/O to all devices and the only way to
 recover
 is to reboot the system.
 That's obviously *not* the root cause.

 It's not normal for an application that isn't even privileged to hang
 all I/O and, subsequently everything on a system.

 This is almost probably a kernel issue and asterisk just does something
 that triggers this bug.

 Regards,
 Faidon


It is possible although I tried with several kernels (2.6.30.[0-9]  
2.6.31+ (never had a crash with earlier versions, I installed asterisk long
ago) but it always used to be 1.4.x until recently..  Nasty bug :\

Justin.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] Bug#557262: 2.6.31+2.6.31.4: XFS - All I/O locks up to D-state after 24-48 hours (sysrq-t+w available) - root cause found = asterisk

[Luki]

 When this bug occurs, it freezes I/O to all devices and the only way to
 recover is to reboot the system.

Are you running asterisk with realtime priority (-p)?

I once managed to take town a box with a dial plan loop; asterisk was
taking to 100% CPU and because it had highest priority, nothing else
would run. Kernel would respond to pings, but that's it. We no longer
use realtime priority for that reson :).

Luki

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] Bug CDR report - dst s ?

[Diana Lopez]

Hello everybody,

I have a question about value dst of cdr table in asteriskcdrdb,  so in
my db I see many registers with letter s in dst field, I found a
opinion that s is the default extension and the all calls enter the
system as s and are then changed as they pass through the dial plan,
so, if you have an IVR and they hang up during the IVR, the call will be
s.   Is it true?

And if it's true there's a solution to the dst put a number or is normal
this case, I want to change s by a value.

Thanks a lot

-- 


Diana López



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] Bug or feature: SIP chanvars not overriden

[Olivier]

Hello,

Using 1.6.2-rc5, my settings include:

[local-phone](!)
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
setvar=accountcode=foo

[168](local-phone)
defaultuser=168
secret=pass168
callerid=John Doe168
mailbox=168
setvar=longcid=01555
setvar=accountcode=bar


CLI sip show peer 168
...
  Variables:
 accountcode = bar
 longcid = 01555
 accountcode = foo


When running, ${SIPPEER(168,chanvar[accountcode])}) is valued to foo
(instead of bar).
Would you rate it as a feature ?

Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] Bug or feature: SIP chanvars not overriden

[Tilghman Lesher]

On Wednesday 11 November 2009 14:23:31 Olivier wrote:
 Hello,

 Using 1.6.2-rc5, my settings include:

 [local-phone](!)
 context=mylocal
 type=friend
 nat=no
 canreinvite=no
 host=dynamic
 qualify=yes
 dtmf=info
 language=fr
 call-limit=5
 subscribecontext=subs
 disallow=all
 allow=alaw
 t38pt_udptl=no
 setvar=accountcode=foo

 [168](local-phone)
 defaultuser=168
 secret=pass168
 callerid=John Doe168
 mailbox=168
 setvar=longcid=01555
 setvar=accountcode=bar


 CLI sip show peer 168
 ...
   Variables:
  accountcode = bar
  longcid = 01555
  accountcode = foo


 When running, ${SIPPEER(168,chanvar[accountcode])}) is valued to foo
 (instead of bar).
 Would you rate it as a feature ?

Neither.  It's a misunderstanding on your part of how this all works.  The
equivalent of your entry above is the context:

[168]
context=mylocal
type=friend
nat=no
canreinvite=no
host=dynamic
qualify=yes
dtmf=info
language=fr
call-limit=5
subscribecontext=subs
disallow=all
allow=alaw
t38pt_udptl=no
setvar=accountcode=foo
defaultuser=168
secret=pass168
callerid=John Doe168
mailbox=168
setvar=longcid=01555
setvar=accountcode=bar

The FIRST value is the value which takes precedence, not the last.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s![swapper:0]

[Lee, John (Sydney)]

 I'd contact Digium - they're really good with providing support - just

 add the following line and dial it:

Thanks Matt for your suggestion.
We despatched a new TE412P card to replace the existing card but the
same problem occurred.  So, I think it is not the Digium card problem.

At the same time, we noticed that the 2nd port (which is configured as a
T1 to connect to a Rhino Channel Bank) was reporting red/rec in zttool.
So, we unplug the 2nd port and the soft lockup problem goes away.
However, doing so means we cannot configure and use the analog channels
from the E1 ISDN line which is connected to port 1 on the TE412P.

I reported the problem to Rhino and the support confidently believed
that the issues are related to the OS and platform and not the Rhino. 

Anyone has any suggestion?

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Luis Morales]

The error on system crash is:

Digum Board: TDM2400P
OS: Debian Lenny 5.02

dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38
KERNEL PANIC NOT SYNCING


There are any imcompatibility on servers based on quadcore ? or dahdi
not have correct support for this model  ?

Regards,



On Mon, Aug 17, 2009 at 2:17 AM, Alex Samada...@samad.com.au wrote:
 On Sat, Aug 15, 2009 at 10:58:07PM +1000, Lee, John (Sydney) wrote:
 I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
 problems since Dec last year. We are using Digium TE412P to connect to

 [snip]

 Pid: 0, comm: swapper
 EIP: 0060:[,C0417911.] CPU: 1
 EIP is at smp_call_function+0x99/0xc3
 EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1)
 EAX: 0002 EBX:  ECX: 0001 EDX: 00fb
 ESI: 0003 EDI:  EBP: c0417ae0 DS: 007B ES: 007b
 CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0]
 stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf]
 panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146

 Have you installed mce - the machine check error logger to see what this
 is, you might get more relevant information from there.


 I have also seen this cpu stuck when my adsl goes down and i get a very
 short lifed ipv6 routing looping - sucks up cpu and doesn't let it intr
 which starves the other cards.

 Alex

 [c040d91e] intel_machine_check+0x0/0x146 [c0403ccf]
 error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200]
 acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9
 ===

 Q1. A strange thing is I could not find this error message in
 /var/log/messages or dmesg. The soft lockup error message can only be
 found on the machine itself.

 Q2. Could it be kernel incompatibility problem? However, we did not ever
 change anything since it was installed.

 Q3. From the error message, how do I know it is a software (kernel?) or
 hardware problem?

 I would appreciate if someone could give me any suggestions.



 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

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


 --
 Quick, you must come with me, you're in great danger!
 Why?
 Because I'll kill you if you dont.
 (Sourcery)

 -BEGIN PGP SIGNATURE-
 Version: GnuPG v1.4.9 (GNU/Linux)

 iEYEARECAAYFAkqHq40ACgkQkZz88chpJ2Od6gCcC9UtTSQwISeqvHq/NO744S8r
 3EMAoOKh/xlGHxBw1MTpYM+2P6dnBypo
 =Hmuq
 -END PGP SIGNATURE-

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

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




-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Kevin P. Fleming]

Luis Morales wrote:
 The error on system crash is:
 
 Digum Board: TDM2400P
 OS: Debian Lenny 5.02
 
 dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38
 KERNEL PANIC NOT SYNCING
 
 
 There are any imcompatibility on servers based on quadcore ? or dahdi
 not have correct support for this model  ?

Are you using HPEC as your software echo canceller? If so, which CPU
flavor of HPEC did you install? It is possible you've installed a
version which isn't compatible with your CPU and it's trying to execute
instructions that your CPU does not support.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Luis Morales]

You rite Kevin,

We enabeled sec echo canceller. I'll be test now and let's know  the results.

Regards,

On Tue, Aug 18, 2009 at 8:47 AM, Kevin P. Flemingkpflem...@digium.com wrote:
 Luis Morales wrote:
 The error on system crash is:

 Digum Board: TDM2400P
 OS: Debian Lenny 5.02
 
 dahdi_ec_chunk 0x3c/0x200 [dahdi] SS:ESP 00068:F747FE38
 KERNEL PANIC NOT SYNCING
 

 There are any imcompatibility on servers based on quadcore ? or dahdi
 not have correct support for this model  ?

 Are you using HPEC as your software echo canceller? If so, which CPU
 flavor of HPEC did you install? It is possible you've installed a
 version which isn't compatible with your CPU and it's trying to execute
 instructions that your CPU does not support.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kpflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

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




-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Kevin P. Fleming]

Luis Morales wrote:
 You rite Kevin,
 
 We enabeled sec echo canceller. I'll be test now and let's know  the results.

SEC is not a good choice. If you are going to try something other than
HPEC, use MG2 or KB1, which are the current best options that are
included with DAHDI. You can also stick with HPEC, but just use a
'generic' CPU flavor instead of a highly-optimized version.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Luis Morales]

Keving,

We use MG2 and KB1, but the best result was on SEC. How i can do to
modify dahsi to include SEC option with generic CPU.

Jhon Lee,

Take a look on this link, there are an option to solve your issue:
http://archives.free.net.ph/message/20080126.111546.a2569851.nl.html



Regards,

On Tue, Aug 18, 2009 at 10:46 AM, Kevin P. Flemingkpflem...@digium.com wrote:
 Luis Morales wrote:
 You rite Kevin,

 We enabeled sec echo canceller. I'll be test now and let's know  the results.

 SEC is not a good choice. If you are going to try something other than
 HPEC, use MG2 or KB1, which are the current best options that are
 included with DAHDI. You can also stick with HPEC, but just use a
 'generic' CPU flavor instead of a highly-optimized version.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kpflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

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




-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Kevin P. Fleming]

Luis Morales wrote:
 Keving,
 
 We use MG2 and KB1, but the best result was on SEC. How i can do to
 modify dahsi to include SEC option with generic CPU.

If you are not using HPEC, then none of this matters; when you use an
echo canceller included with DAHDI, it's compiled for your CPU type and
this is not an issue.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Alex Samad]

On Sat, Aug 15, 2009 at 10:58:07PM +1000, Lee, John (Sydney) wrote:
 I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
 problems since Dec last year. We are using Digium TE412P to connect to

[snip]

 Pid: 0, comm: swapper
 EIP: 0060:[,C0417911.] CPU: 1
 EIP is at smp_call_function+0x99/0xc3
 EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1)
 EAX: 0002 EBX:  ECX: 0001 EDX: 00fb
 ESI: 0003 EDI:  EBP: c0417ae0 DS: 007B ES: 007b
 CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0]
 stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf]
 panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146

Have you installed mce - the machine check error logger to see what this
is, you might get more relevant information from there.


I have also seen this cpu stuck when my adsl goes down and i get a very
short lifed ipv6 routing looping - sucks up cpu and doesn't let it intr
which starves the other cards.

Alex

 [c040d91e] intel_machine_check+0x0/0x146 [c0403ccf]
 error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200]
 acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9
 ===
 
 Q1. A strange thing is I could not find this error message in
 /var/log/messages or dmesg. The soft lockup error message can only be
 found on the machine itself.
 
 Q2. Could it be kernel incompatibility problem? However, we did not ever
 change anything since it was installed.
 
 Q3. From the error message, how do I know it is a software (kernel?) or
 hardware problem?
 
 I would appreciate if someone could give me any suggestions.
 
 
 
 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
 

-- 
Quick, you must come with me, you're in great danger!
Why?
Because I'll kill you if you dont.
(Sourcery)


signature.asc
Description: Digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Lee, John (Sydney)]

I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
problems since Dec last year. We are using Digium TE412P to connect to
an E1 ISDN line. Since Dec last year, we did not add or delete any
software or hardware. We also did not do any yum update.

The linux kernel is 2.6.18-92.1.22.el5

Last week, the users reported that people from outside could not dial in
but users can dial out. We rebooted the box and everything was fine.

Suddenly, starting this week, the box froze several times a day with a
BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on
the console. Before it freezes, I can see a continuous stream of error
message ...
timing source auto card 0!
timing source auto card 0!
timing source auto card 0!
timing source auto card 0!
...
coming up on the machine.

We rebooted and it became okay for a few hours and we had to reboot it
again in order to clear the problem.

BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
Pid: 0, comm: swapper
EIP: 0060:[,C0417911.] CPU: 1
EIP is at smp_call_function+0x99/0xc3
EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1)
EAX: 0002 EBX:  ECX: 0001 EDX: 00fb
ESI: 0003 EDI:  EBP: c0417ae0 DS: 007B ES: 007b
CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0]
stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf]
panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146
[c040d91e] intel_machine_check+0x0/0x146 [c0403ccf]
error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200]
acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9
===

Q1. A strange thing is I could not find this error message in
/var/log/messages or dmesg. The soft lockup error message can only be
found on the machine itself.

Q2. Could it be kernel incompatibility problem? However, we did not ever
change anything since it was installed.

Q3. From the error message, how do I know it is a software (kernel?) or
hardware problem?

I would appreciate if someone could give me any suggestions.



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Lee, John (Sydney)]

I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
problems since Dec last year. We are using Digium TE412P to connect to
an E1 ISDN line. Since Dec last year, we did not add or delete any
software or hardware. We also did not do any yum update.

The linux kernel is 2.6.18-92.1.22.el5

Last week, the users reported that people from outside could not dial in
but users can dial out. We rebooted the box and everything was fine.

Suddenly, starting this week, the box froze several times a day with a
BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on
the console. Before it freezes, I can see a continuous stream of error
message 
...
timing source auto card 0!
timing source auto card 0!
timing source auto card 0!
timing source auto card 0!
...
coming up on the machine.

We rebooted and it became okay for a few hours and we had to reboot it
again in order to clear the problem.

BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
Pid: 0, comm: swapper
EIP: 0060:[,C0417911.] CPU: 1
EIP is at smp_call_function+0x99/0xc3
EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1)
EAX: 0002 EBX:  ECX: 0001 EDX: 00fb
ESI: 0003 EDI:  EBP: c0417ae0 DS: 007B ES: 007b
CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0
[c0417ae0] stop_this_cpu+0x0/0x33
[c041794e] smp_send_stop+0x13/0x1c
[c0425bcf] panic+0x4c/0x16d
[c040da17] intel_machine_check+0xf9/0x146
[c040d91e] intel_machine_check+0x0/0x146
[c0403ccf] error_code+0x39/0x40
[c0403ccf] mwait_idel+0x25/0x38
[c0522200] acpi_processor_idle+0x154/0x3b4
[c0403c90] cpu_idle+0x9f/0xb9
===

Q1. A strange thing is I could not find this error message in
/var/log/messages or dmesg. The soft lockup error message can only be
found on the machine itself.

Q2. Could it be kernel incompatibility problem? However, we did not ever
change anything since it was installed.

Q3. From the error message, how do I know it is a software (kernel?) or
hardware problem?

I would appreciate if someone could give me any suggestions.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Matt Riddell]

On 16/08/09 12:58 AM, Lee, John (Sydney) wrote:
 I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
 problems since Dec last year. We are using Digium TE412P to connect to
 an E1 ISDN line. Since Dec last year, we did not add or delete any
 software or hardware. We also did not do any yum update.

 The linux kernel is 2.6.18-92.1.22.el5

 Last week, the users reported that people from outside could not dial in
 but users can dial out. We rebooted the box and everything was fine.

 Suddenly, starting this week, the box froze several times a day with a
 BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on
 the console. Before it freezes, I can see a continuous stream of error
 message ...

I'd contact Digium - they're really good with providing support - just 
add the following line and dial it:

Dial(IAX2/gu...@pbx.digium.com/s...@default)

-- 
Cheers,

Matt Riddell
Director
___

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]

[Luis Morales]

I have the same trouble but on hp server. In my case the digium board
used is : TE121P.  My server is an ML150 G5 on ubuntu 8.04.2.TLS.

I'll be appreciate if you can solve it.

Regards,


On Sun, Aug 16, 2009 at 6:36 PM, Matt Riddellli...@venturevoip.com wrote:
 On 16/08/09 12:58 AM, Lee, John (Sydney) wrote:
 I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no
 problems since Dec last year. We are using Digium TE412P to connect to
 an E1 ISDN line. Since Dec last year, we did not add or delete any
 software or hardware. We also did not do any yum update.

 The linux kernel is 2.6.18-92.1.22.el5

 Last week, the users reported that people from outside could not dial in
 but users can dial out. We rebooted the box and everything was fine.

 Suddenly, starting this week, the box froze several times a day with a
 BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on
 the console. Before it freezes, I can see a continuous stream of error
 message ...

 I'd contact Digium - they're really good with providing support - just
 add the following line and dial it:

 Dial(IAX2/gu...@pbx.digium.com/s...@default)

 --
 Cheers,

 Matt Riddell
 Director
 ___

 http://www.venturevoip.com/news.php (Daily Asterisk News)
 http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
 http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

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




-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Bug or Not?

[Danny Nicholas]

Hi gang,

 When I try to park a call using blind-transfer (#1),  the
caller hears the lot instead of the transferring party.  Attended transfer
and blind transfer from the phone buttons (Polycom 501) work fine, so this
isn't a showstopper, just a WHY??.  Thanks for you attention.

 

Danny Nicholas

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] Bug or Not?

[Paul Hales]

'One touch park' was designed to work around this issue.

PaulH


Danny Nicholas wrote:

 Hi gang,

 When I try to park a call using blind-transfer (#1), the caller hears
 the lot instead of the transferring party. Attended transfer and blind
 transfer from the phone buttons (Polycom 501) work fine, so this isn’t
 a showstopper, just a “WHY??”. Thanks for you attention.

 Danny Nicholas

 

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 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 --

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

[asterisk-users] BUG in Asterisk 1.6.1.0 and issue in DAHDI 2.1.0.4

[Nestor A. Diaz]

Starting playing with asterisk 1.6.1.0 i found the following problems:

In the cdr_pgsql, the sql statement is wrong:

2009-06-25 12:17:01 COT LOG:  statement: INSERT INTO cdr
(accountcode,calldate,src,clid,dst,src,dcontext,dst,clid,dcontext,channel,channel,dstchannel,lastapp,lastapp,lastdata,lastdata,duration,start,billsec,answer,disposition,end,duration,amaflags,billsec,accountcode,disposition,uniqueid,amaflags,userfield,userfield,uniqueid)
VALUES ('4868','2009-06-25 12:14:11','unknown','bga-gw-1-4
unknown','4','unknown','from-bga-gw','4','bga-gw-1-4
unknown','from-bga-gw','SIP/bga-gw-1-4-09be9960','SIP/bga-gw-1-4-09be9960','DAHDI/5-1','Dial','Dial','DAHDI/g11/4868','DAHDI/g11/4868',170,'2009-06-25
12:14:11',170,'2009-06-25 12:14:11','ANSWERED','2009-06-25
12:17:01',170,3,170,'4868',8,'1245950051.272',3,'4','4','1245950051.272')
2009-06-25 12:17:01 COT ERROR:  column src specified more than once

and asterisk shows:

[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:309 pgsql_log: Failed to
insert call detail record into database!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:310 pgsql_log: Reason:
ERROR:  column src specified more than once

[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:311 pgsql_log: Connection
may have been lost... attempting to reconnect.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:314 pgsql_log: Connection
reestablished.
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:320 pgsql_log: HARD ERROR!
Attempted reconnection failed.  DROPPING CALL RECORD!
[Jun 25 12:18:07] ERROR[15544]: cdr_pgsql.c:321 pgsql_log: Reason:
ERROR:  column src specified more than once

Also in chan_dahdi i got the following:

[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: k
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: o
[Jun 25 12:19:04] WARNING[15757]: chan_dahdi.c:3090 dahdi_call:
Unrecognized prilocaldialplan NPI modifier: w

Slds.

-- 
Nestor A. Diaz
Ingeniero de Sistemas
Tel. +57 1-600-5490 x 211
Cel. +57 316-227-3593
Tel. SIP: sip:2...@tiendalinux.com
Email/MSN: nes...@tiendalinux.com
http://www.tiendalinux.com/
Bogota, Colombia 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] Bug or feature : how to customize SIP REFER from dialplan

[Olivier]

Hi,

I've been editing my dialplan to launch custom instructions anytime a SIP
REFER-based transfer occurs.

The only hook I could find is catching an hangup event which is tied to a
Zombie channel
(ie a channel named like SIP/1234-vhvebjvnvZOMBIE).

Is this a feature or a bug ?
In other words, do you think :
- it shouldn't be possible at all to hook custom instructions for SIP
REFER-based transfer occurs (then I obviously found a bug),
- catching ZOMBIE channel hangup is the way to hook custom instructions for
a SIP REFER-based transfer.

Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?

[Olivier]

I filed https://issues.asterisk.org/view.php?id=15202
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] Bug or feature in 1.6.1 (Was: How to register with TCP transport) ?

[Olivier]

Hi,

Digging on this case :

2009/5/26 Olivier oza-4...@myamail.com

 Hi,

 In my sip.conf, I've got :
 [general](+)
 ;   
 register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
 
 register=trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129

 When I'm using the TCP line instead of the other, I've got :
 [May 26 17:58:42] NOTICE[2859]: chan_sip.c:20169 sip_parse_host: '/' is not
 a valid port number on line 25 of sip.conf. using default.
 [May 26 17:58:42] WARNING[2859]: chan_sip.c:6560 sip_register: Format for
 registration is
 [transport://]user[:secret[:authuse...@domain[:port][/extension][~expiry] at
 line 25


 Is this 
 register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
 statement correct ?

 Regards



I read in chan_sip.c that block inside sip_register :

   /* split [/contact][~expiry] */
expire = strchr(buf, '~');
if (expire)
*expire++ = '\0';
callback = strrchr(buf, '/');// My comment: contact is
search at the end of input register line
if (callback)
*callback++ = '\0';
if (ast_strlen_zero(callback))
callback = s;

sip_parse_host(buf, lineno, username, portnum, transport);

Given an input line such as register=tcp://
trunk4ipbx:passw...@192.168.100.129 trunk4ipbx%3apassw...@192.168.100.129,
register line is truncated as the last occurence of '/' is the tcp://
string.
When commenting out this callback = strrchr(buf, '/'); , input line
register=tcp://trunk4ipbx:passw...@192.168.100.129trunk4ipbx%3apassw...@192.168.100.129
seems to be processed appropriately.

My question is is this legal to input register lines without any /contact
field ?
If positive, then there is a bug is 1.6.1.
If negative, would you agree to have a more appropriate logging than
sip_parse_host: '/' is not a valid port number ... ?

Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] bug(?) bandwidth problem

[David fire]

hi
i am using asterisk 1.4.22
ubuntu 8.4

i have two Ethernet one for ssh and other one only for voip calls

when i start a call using originate in the manager or the cli
in the voip Ethernet i get something like 4Mbits/sec of traffic only  1 G711
call.
if i start the call using a soft phone everything is normal.

any idea?
thanks


-- 
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

[asterisk-users] bug 14153 and svn checkout.

[Jerry Geis]

I just did an SVN check out and the fix for bug 14153 was not included 
in the SVN checkout.
Is there something special I need to issue in the SVN checkout to get it?

Jerry

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] bug 14153 and svn checkout.

[Jerry Geis]

Jerry Geis wrote:
 I just did an SVN check out and the fix for bug 14153 was not included 
 in the SVN checkout.
 Is there something special I need to issue in the SVN checkout to get it?

 Jerry

I did not include the command I used.
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk


using this command and looking at channels/chan_alsa.c the fixes are not 
included.
How do I correctly checkout an svn that includes the fix?

Jerry

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] bug 14153 and svn checkout.

[Philipp Kempgen]

Jerry Geis schrieb:
 Jerry Geis wrote:
 I just did an SVN check out and the fix for bug 14153 was not included 
 in the SVN checkout.
 Is there something special I need to issue in the SVN checkout to get it?

 I did not include the command I used.
 svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
 
 
 using this command and looking at channels/chan_alsa.c the fixes are not 
 included.

Looks like http://bugs.digium.com/view.php?id=14153 was closed
by accident (typo in the commit message).

I think tilghman meant to close
http://bugs.digium.com/view.php?id=14151

http://bugs.digium.com/view.php?id=14153 needs to be reopened.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] bug 14153 and svn checkout.

[Mark Michelson]

Philipp Kempgen wrote:
 Jerry Geis schrieb:
 Jerry Geis wrote:
 I just did an SVN check out and the fix for bug 14153 was not included 
 in the SVN checkout.
 Is there something special I need to issue in the SVN checkout to get it?
 
 I did not include the command I used.
 svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk


 using this command and looking at channels/chan_alsa.c the fixes are not 
 included.
 
 Looks like http://bugs.digium.com/view.php?id=14153 was closed
 by accident (typo in the commit message).
 
 I think tilghman meant to close
 http://bugs.digium.com/view.php?id=14151
 
 http://bugs.digium.com/view.php?id=14153 needs to be reopened.
 
 
Philipp Kempgen
 

I re-opened this bug. Thanks for bringing this up.

Mark Michelson

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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

Re: [asterisk-users] bug 14153 and svn checkout.

[Jerry Geis]

Jerry Geis wrote:

Jerry Geis wrote:
I just did an SVN check out and the fix for bug 14153 was not 
included in the SVN checkout.
Is there something special I need to issue in the SVN checkout to get 
it?


Jerry


I did not include the command I used.
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk


using this command and looking at channels/chan_alsa.c the fixes are 
not included.

How do I correctly checkout an svn that includes the fix?

Jerry


Looks like I was executing the wrong command:

|svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 


was the command I needed.
I now get the changes for bug 14153.

Jerry
|


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

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