[asterisk-users] core show translation - difference in Asterisk Versions

[Salman Zafar]

Hello All,
   I was wondering if somebody could elaborate the change in
translation of codecs specifically the amount of time increased in Asterisk
11. For example

*Asterisk 11*
***alaw **speex *
*gsm **15000 **15000 *
*ulaw9150   15000*
* *
*Asterisk 1.6.x*
***alaw **speex *
*gsm **2 12002 *
*ulaw1 12002*

I did recalculate the translation for 60 or 1 seconds but nothing changes
on Asterisk 11 (VM, Cloud or even physical machine). Is it slin?, adding
this overhead or there is something I am overlooking?.
*
*
*Asterisk 11.0.1 => core show translation **(in microseconds)*

*gsm  ulaw  alaw  g726 adpcm  slin lpc10  g729 speex speex16
 ilbc g726aal2  g722 slin16 testlaw speex32 slin12 slin24 slin32 slin44
slin48 slin96 slin192*
  *gsm *- 15000 *15000 *15000 15000  9000 15000 15000 *15000   *23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *ulaw *15000 -  9150 15000 15000  9000 15000 15000 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *alaw *15000  9150 - 15000 15000  9000 15000 15000 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *g726 *15000 15000 15000 - 15000  9000 15000 15000 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
*adpcm *15000 15000 15000 15000 -  9000 15000 15000 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *slin  *6000  6000  6000  6000  6000 -  6000  6000  6000   14000
 6000 6000  8250   80006000   14000   8000   8000   8000   8000
8000   80008000
*lpc10 *15000 15000 15000 15000 15000  9000 - 15000 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *g729 *15000 15000 15000 15000 15000  9000 15000 - 15000   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
*speex *15000 15000 15000 15000 15000  9000 15000 15000 -   23000
1500015000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
  *speex16 *23500 23500 23500 23500 23500 17500 23500 23500 23500   -
2350023500 15000   9000   23500   23000  17500  17000  17000  17000
 17000  17000   17000
 *ilbc *15000 15000 15000 15000 15000  9000 15000 15000 15000   23000
  -15000 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *g726aal2 *15000 15000 15000 15000 15000  9000 15000 15000 15000   23000
15000- 17250  17000   15000   23000  17000  17000  17000  17000
 17000  17000   17000
 *g722 *15600 15600 15600 15600 15600  9600 15600 15600 15600   15000
1560015600 -   9000   15600   23000  17500  17000  17000  17000
 17000  17000   17000
   *slin16 *14500 14500 14500 14500 14500  8500 14500 14500 145006000
1450014500  6000  -   14500   14000   8500   8000   8000   8000
8000   80008000
  *testlaw *15000 15000 15000 15000 15000  9000 15000 15000 15000   23000
1500015000 17250  17000   -   23000  17000  17000  17000  17000
 17000  17000   17000
  *speex32 *23500 23500 23500 23500 23500 17500 23500 23500 23500   23500
2350023500 23500  17500   23500   -  17500  17500   9000  17000
 17000  17000   17000
   *slin12 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14000
1450014500 14000   8000   14500   14000  -   8000   8000   8000
8000   80008000
   *slin24 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   14500   14000   8500  -   8000   8000
8000   80008000
   *slin32 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   145006000   8500   8500  -   8000
8000   80008000
   *slin44 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   14500   14500   8500   8500   8500  -
8000   80008000
   *slin48 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   14500   14500   8500   8500   8500   8500
 -   80008000
   *slin96 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   14500   14500   8500   8500   8500   8500
8500  -8000
  *slin192 *14500 14500 14500 14500 14500  8500 14500 14500 14500   14500
1450014500 14500   8500   14500   14500   8500   8500   8500   8500
8500   8500   -

*Asterisk 1.6.2.x => core show translation **(in microseconds)*

   *g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex
 ilbc  g726  g722 siren7 siren14 slin16*
 *g723* - - - -- - - - - -
- - -  -   -  -
  *gsm *- - 2 *2 *400

Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

[Face]

I upgrading to 11 because I want to use the "MessageSend" command from the
AMI, ver 10 dose not have "MessageSend" In the list of
commands. Unfortunately I remove  ver 11 and I dont think I can provide the
information you asked.


On Tue, Nov 20, 2012 at 4:46 PM, Joshua Colp  wrote:

> Face wrote:
>
>>
>> Well, thanks for responding. I went back to 10.10.0 and things seem to
>> be working fine now!
>>
>
> This is certainly good to know but I'd like to know why upgrading to 11
> did not seem to work for you. This is the first case since it's been out
> where it doesn't appear to have been smooth. Would you be willing to
> provide the information I asked about from a running 11 instance?
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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Re: [asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE

[upendra]

Hi ,


it implies that under this CONFIG_VOICEBUS_ECREFERENCE code inside this
condition will never execute.
if we need it should be defined .


Regards
Upendra

On Tue, Nov 20, 2012 at 8:55 PM, Shaun Ruffell  wrote:

> On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote:
> > Hi,
> >
> > just going through the code i found that this
>  CONFIG_VOICEBUS_ECREFERENCE
> > undef in the voicebus then how it will defined to run the echo cancell on
> > the respective  drievers wctdm24xxp ??
> >  explain how this  CONFIG_VOICEBUS_ECREFERENCE  enabled and where it is
> > enabled while run time.
>
> It's a compile time option that was in added to the code for testing
> in certain environments [1]. It doesn't enable / disable the
> echocanceler, but instead changes the default processing to provide
> a more accurate reference signal when latency grows large.
>
> [1] http://svnview.digium.com/svn/dahdi?view=revision&revision=9144
>
> If you would like to use it for some reason, in
> drivers/dahdi/voicebus.h change "#undef CONFIG_VOICEBUS_ECREFERENCE"
> to "#define CONFIG_VOICEBUS_ECREFERENCE".
>
> In practice, I've found that it's generally better to fix what on
> the system is causing latency to grow so large instead of enabling
> that option.
>
> Cheers,
> Shaun
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] watchdog like functions

[Carlos Alvarez]

Switching to SIP is likely your best solution.  IAX is buggy.  Always has
been, and I'll bet always will be.


On Tue, Nov 20, 2012 at 7:34 PM, asterisk asterisk wrote:

> I wish to ask if there is way to keep IAX trunk connection up. I have a
> small server on Xen VPS but notice that my IAX trunk drops after some time.
>
> I understand there is cron job to function as sip watchdog.
>
> My asterisk is 11.0.1
>
> Thanks for suggestions.
>
> CK
>
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-- 
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TelEvolve
602-889-3003
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[asterisk-users] watchdog like functions

[asterisk asterisk]

I wish to ask if there is way to keep IAX trunk connection up. I have a
small server on Xen VPS but notice that my IAX trunk drops after some time.

I understand there is cron job to function as sip watchdog.

My asterisk is 11.0.1

Thanks for suggestions.

CK
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Re: [asterisk-users] If would possible use a custom function in Asterisk Dialplan

[Andrew White]

Hey Longst,

I'd recommend having a look into the LUA support Asterisk offers for its 
dialplans or AGI. These are the only realistic ways to add functions, unless 
you want to write your own C module and compile it in. Adhearsion is an option 
as well, if you are proficient with ruby.

Cheers,

Andrew

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Tuesday, 20 November 2012 2:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] If would possible use a custom function in Asterisk 
Dialplan

Hello,

If would be possible to use a "function concept" in side of Asterisk DialPlan 

For example:

I have following logic in my dial plan remove country code a add an "0" before 
the rest of the numbers 


exten => _X.,1, NoOp( call ID ${CALLERID(num)}  exten: ${EXTEN})) ; remove my 
country code exten => _X.,n, 
GotoIf($["${CALLERID(num):0:4}"="${country-code}"]?international-format:national-format)
exten => _X.,n(international-format), Set(CALLERID(num)=0${CALLERID(num):4})
exten => _X.,n(national-format), NoOp(call ID: ${CALLERID(num)} exten: 
${EXTEN}))

Do you think if would be possible that I could write a function something like 
"REMOVEMYCOUNTRYCODE(${NUM})" with a return value of a number with out country 
code and with an "0" add in front of the rest of the numbers.

like 

exten => _X.,1, NoOp( call ID ${CALLERID(num)}  exten: ${EXTEN})) ; remove my 
county code exten => _X.n, 
Set(CALLERID(num=REMOVEMYCOUNTRYCODE(${CALLERID(num)} )); 

then I have to define this function in someway ..

I am trying to googling for a while but I did not find any idea to achieve this 
task. 

I would appreciate if someone have an idea...

Thanks for your time in advance.


longst


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Re: [asterisk-users] No more connections allowed.

[Danny Nicholas]

My suggestion would be to issue a "core restart when convenient" around
midnight, assuming your installation doesn't do 24/7 calls.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonson Player
Sent: Tuesday, November 20, 2012 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No more connections allowed.

 

Hello Danny,

 

Could you tell me how can i put time out at execution of remote commands
with asterisk -rx "show sip channels".

I think that is my problem... after i execute asterisk -rx commands
something remain stalled and somehow i think that could block my asterisk...

I mean all new connections couldn't be made anymore and the old active peers
is nonfunctional and logged off, and the strangest thing is that i go on
asterisk -rvc and i tap sip show peers

i seen old active peers logged in... is clear that asterisk is freezed
somehow and i need some workaround at this situation.

 

Thank you.

On Tue, Nov 20, 2012 at 4:50 PM, Danny Nicholas  wrote:


You've exceeded the allowed maximum number of simultaneous remote console
connections (128).  While it may be a bit aggressive to have it kill all
current connected consoles, its also a bit excessive to have
128 connected remote consoles.

While the behaviour may not be entirely desirable, this isn't so much a bug
as a limitation of the system.

--

Why would you need 128 simultaneous remote consoles?  IMO the number should
be something like 16 and a timeout option needs to be added so you don't
have unattended consoles consuming resources and opening holes.



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Re: [asterisk-users] No more connections allowed.

[Eric Wieling]

No, you don't need your script for this.

Prevent attacks by using fail2ban to block brute force attacks using iptables, 
securing your server at the OS level, and NEVER EVER EVER let leave the web GUI 
for FreePBX open to the internet.  I'm sure others have more suggestions.

Over the years 100% of successful attacks against our Asterisk/FreePBX servers 
(we have 20 or 30 servers) used the FreePBX GUI to get access to the server.   
We use very complex passwords for SIP accounts and use a non-default password 
for the Polycom provisioning user.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonson Player
Sent: Tuesday, November 20, 2012 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] No more connections allowed.

[snip]

What is your advice? I need this analyzing script to monitor when i have 
strange rise of use channels to prevent attacks or brute force.

 


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Re: [asterisk-users] No more connections allowed.

[Jonson Player]

Hello Matthew,

Could I rise with some option the number of simultaneous console? I don't
have simultaneous console but is good to know in case i didn't get any
other workaround to fix this problem.

Thank you.

On Tue, Nov 20, 2012 at 3:56 PM, Matthew Jordan  wrote:

> On 11/20/2012 03:32 AM, Jonson Player wrote:
> > Hello,
> >
> > I have strange situation with asterisk 1.8.18.0 , randomly i got this
> > message in cli:
> >
> > "WARNING[15925] asterisk.c: No more connections allowed"
> >
> > All connections freeze and all extensions doesn't work anymore. Is any
> > bug or is any setting that can solve this problem?
> >
> > Thank you.
> >
>
> You've exceeded the allowed maximum number of simultaneous remote
> console connections (128).  While it may be a bit aggressive to have it
> kill all current connected consoles, its also a bit excessive to have
> 128 connected remote consoles.
>
> While the behaviour may not be entirely desirable, this isn't so much a
> bug as a limitation of the system.
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>
>
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Re: [asterisk-users] No more connections allowed.

[Jonson Player]

Hello Danny,

Could you tell me how can i put time out at execution of remote commands
with asterisk -rx "show sip channels".
I think that is my problem... after i execute asterisk -rx commands
something remain stalled and somehow i think that could block my asterisk...
I mean all new connections couldn't be made anymore and the old active
peers is nonfunctional and logged off, and the strangest thing is that i go
on asterisk -rvc and i tap sip show peers
i seen old active peers logged in... is clear that asterisk is freezed
somehow and i need some workaround at this situation.

Thank you.

On Tue, Nov 20, 2012 at 4:50 PM, Danny Nicholas  wrote:

>
> You've exceeded the allowed maximum number of simultaneous remote console
> connections (128).  While it may be a bit aggressive to have it kill all
> current connected consoles, its also a bit excessive to have
> 128 connected remote consoles.
>
> While the behaviour may not be entirely desirable, this isn't so much a bug
> as a limitation of the system.
>
> --
> Why would you need 128 simultaneous remote consoles?  IMO the number should
> be something like 16 and a timeout option needs to be added so you don't
> have unattended consoles consuming resources and opening holes.
>
>
> --
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Re: [asterisk-users] No more connections allowed.

[Jonson Player]

Hello Josua,

Thank you for this answers:

First of all, yes i run in crontab at 15 min an analyzing script which
collect show sip channels with asterisk -rx . This could be my problem...
I think that this commands could remain stalled and doesn't logout after
execution of command.

A friend of main told me that i could put in asterisk init script the
following command: ulimit -HSn 135535

You think is useful?

Another abortion could be to make something with manager /AMI scripts like
that: http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html

What is your advice? I need this analyzing script to monitor when i have
strange rise of use channels to prevent attacks or brute force.



On Tue, Nov 20, 2012 at 3:44 PM, Joshua Colp  wrote:

> Jonson Player wrote:
>
>> Hello,
>>
>
> Hola,
>
>
>  I have strange situation with asterisk 1.8.18.0 , randomly i got this
>> message in cli:
>>
>> "WARNING[15925] asterisk.c: No more connections allowed"
>>
>
> This message is output when the number of Asterisk consoles (asterisk -r
> instances) has reached the limit. This limit is 128 by default which is
> quite a lot. Are you doing something that would cause a lot? Could they be
> hanging around by mistake?
>
>
>  All connections freeze and all extensions doesn't work anymore. Is any
>> bug or is any setting that can solve this problem?
>>
>
> This definitely shouldn't happen but it would be useful to know exactly
> what you are doing with the system. Answering my questions above is a good
> start.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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>   
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Re: [asterisk-users] Simultaneous caller/callee hangup; hangup extensions execute only once; unable to determine if destination channel up

[Richard Mudgett]

> This is a question regarding whether there's any way within hangup
> extensions to determine whether the caller or callee leg (or both) of
> a
> bridged call has hung up.  The test case I have is running under
> Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and
> also
> 1.6.2.18).
> 
> Within the dialplan, the Dial() application with the "F" flag, so
> that
> once the caller hangs up, the dialplan jumps to a new priority which
> enables the called party to enter some digits which describe the
> outcome
> of the call.  Also, the "g" flag is used to attempt to continue
> execution of the dialplan if the called party hangs up.
> 
> Minimally, the dialplan is covered by the following:
> 
> [test]
> exten => _1000,1,Set(_CALLER_HUNGUP=false)
> exten => _1000,2,Set(_CALLEE_HUNGUP=false)
> exten => _1000,3,Dial(SIP/${EXTEN},60,CgF(test^1000^10))
> exten => _1000,4,Set(_CALLEE_HUNGUP=true)
> 
> exten => _1000,10,Set(_CALLER_HUNGUP=true)
> exten => _1000,11,AGI(afterCallWork.agi)
> 
> exten => h,1,NoOp(${CALLER_HUNGUP})
> exten => h,2,NoOp(${CALLEE_HUNGUP})
> exten => h,3,AGI(postCall.agi)
> 
> 
> Normally, the hangup extensions execute twice: once when the caller
> hangs up, then once more when the called party hangs up, either
> during
> or after the execution of afterCallWork.agi.  This second call is
> important so that clean up can be performed.
> 
> However, if the two parties hang up simultaneously (or within a
> split-second of each other), I often see only one execution of the
> hangup extensions.  Stranger still, the hangups can occur so close to
> each other that execution of the hangup extension occurs without the
> either the priority 4 or priority 10 steps being executed (it can be
> difficult replicate this, but inserting a Wait(1) call at priority 4
> and
> another at priority 10 can help here).
> 
> In such cases, I see the output from the two NoOps as false and
> false.
> (This is difficult to replicate because of the precise timing it
> requires - it is easy if you insert Wait(1) at priority 4 and 10, but
> whether this is valid or not is debatable.  I can replicate this
> issue
> with just the dialplan above on a slowish server).
> 
> So I need to be able to query the status of the other channel from
> within postCall.agi, because if both parties have hung up, I may only
> get one execution of the hangup extensions, and I can go ahead and
> perform the cleanup.  Is this possible?  I've tried calling CHANNEL
> STATUS for the destination channel within postCall.agi, but even when
> the destination channel is definitely still up, the call returns an
> error "511 Command Not Permitted on a dead channel" (presumably
> because
> the current (caller) channel has hung up).
> 
> I can't find anything that I can use within the execution of the
> hangup
> extensions for the caller to determine whether the destination
> channel
> is still up.
> 
> Is it a bug that I only get the one call to the hangup extensions
> when
> both caller and callee channels hangup so close to each other that
> neither the "F" nor "g" flags have the desired effect?

No.  I don't see this as a bug.  Priority 4 and 10 can only execute
while the channel is not hung up.  This is normal dialplan execution.
Only the h exten can execute on a hung up channel.  Since both
channels are hanging up at the same time, neither priority 4 nor 10
are able to get executed.

The new pre-dial and hangup handler features in Asterisk 11 would
be a solution to your problem.  Otherwise, I don't really see a
solution without rethinking your post call processing.

Richard

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Re: [asterisk-users] leading ghost 0

[Alex Kauffmann]

On 11/20/2012 8:03 AM, gincantalupo wrote:

Hi Leandro,

I'm sure nobody has added something... tried prilocaldialplan and
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see
the 0 inside PRI frame with intense debug or it is hidden?

Yes...the technician did it...there is only one cable.

Maybe it is the socket circuitry that has something wrong but I do not
know ho to check.

Asap I'll be on site I'll do more testing.

Thank you

Giorgio

On 11/20/2012 01:13 PM, Leandro Dardini wrote:

That is a real mistery! I like a lots these cases when all seems not
working despite all being correctly configured, but you know first or
later you'll find the answer.

From your website, it seems you are selling/renting PBX based on
asterisk, so you can be sure nobody has messed with the asterisk or
dahdi source code adding a zero... I am sure you have already tried
with a brand new server.

Have you checked the pridialplan and prilocaldialplan setting?

If I was in your shoes, I'll get another server, with a PRI configured
as master and hook it at your PBX to really check if the zero is sent.

Does the technician try to make phone calls from the same network
cable you are using?

Leandro


2012/11/20 gincantalupo mailto:gincantal...@fgasoftware.com>>

Hi Leandro,

thanks for your answer.

I already have tried those parameters but without any positive result.

The telco technician has tried the line with its machine and it
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle that adds
the zero but do not know what it is.
Strange is the fact that you can call some numbers with or without
the prefix zero...
Moreover we had no problem with the previous telco (fastweb).

So we can only call PTSN numbersnot mobile phones.

Giorgio


On 11/20/2012 11:12 AM, Leandro Dardini wrote:

2012/11/20 gincantalupo mailto:gincantal...@fgasoftware.com>>

Hi all,

I have problems dialling out because my new telco (the
previous gave no problems) tells me my PBX adds a leading 0
and that's why I cannot dial out (but I can receive calls).

I make a small extensions.conf as a test:

exten => 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten => 666,1,Dial(DAHDI/g1/0233xx)
and
exten => 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside frames

How can I really be SURE Asterisk is not adding some leading
zero?

Thank you.

Giorgio.


I have never heard of a way to automatically add digits when
using PRI, however can you check your chan_dahdi.conf about the
following lines:

internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in
Italy, every provider has PRI configured in its own way, some
time even the same provider is configuring PRI lines in multiple
times, but often the problems are on receiving the calls (like
calls with and without the area code, with or without the leading
zero, etc. etc.)

Leandro


--


The prilocaldialplan parameter is for inbound so you would have seen no 
changes.  Did you try:


pridialplan=unknown

Did you restart dahdi and asterisk after the changes?

Alex


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[asterisk-users] Simultaneous caller/callee hangup; hangup extensions execute only once; unable to determine if destination channel up

[John Hurst]

Hello

This is a question regarding whether there's any way within hangup 
extensions to determine whether the caller or callee leg (or both) of a 
bridged call has hung up.  The test case I have is running under 
Asterisk 1.8.17.0, but the behaviour is observed in 1.8.18.0 (and also 
1.6.2.18).


Within the dialplan, the Dial() application with the "F" flag, so that 
once the caller hangs up, the dialplan jumps to a new priority which 
enables the called party to enter some digits which describe the outcome 
of the call.  Also, the "g" flag is used to attempt to continue 
execution of the dialplan if the called party hangs up.


Minimally, the dialplan is covered by the following:

[test]
exten => _1000,1,Set(_CALLER_HUNGUP=false)
exten => _1000,2,Set(_CALLEE_HUNGUP=false)
exten => _1000,3,Dial(SIP/${EXTEN},60,CgF(test^1000^10))
exten => _1000,4,Set(_CALLEE_HUNGUP=true)

exten => _1000,10,Set(_CALLER_HUNGUP=true)
exten => _1000,11,AGI(afterCallWork.agi)

exten => h,1,NoOp(${CALLER_HUNGUP})
exten => h,2,NoOp(${CALLEE_HUNGUP})
exten => h,3,AGI(postCall.agi)


Normally, the hangup extensions execute twice: once when the caller 
hangs up, then once more when the called party hangs up, either during 
or after the execution of afterCallWork.agi.  This second call is 
important so that clean up can be performed.


However, if the two parties hang up simultaneously (or within a 
split-second of each other), I often see only one execution of the 
hangup extensions.  Stranger still, the hangups can occur so close to 
each other that execution of the hangup extension occurs without the 
either the priority 4 or priority 10 steps being executed (it can be 
difficult replicate this, but inserting a Wait(1) call at priority 4 and 
another at priority 10 can help here).


In such cases, I see the output from the two NoOps as false and false.  
(This is difficult to replicate because of the precise timing it 
requires - it is easy if you insert Wait(1) at priority 4 and 10, but 
whether this is valid or not is debatable.  I can replicate this issue 
with just the dialplan above on a slowish server).


So I need to be able to query the status of the other channel from 
within postCall.agi, because if both parties have hung up, I may only 
get one execution of the hangup extensions, and I can go ahead and 
perform the cleanup.  Is this possible?  I've tried calling CHANNEL 
STATUS for the destination channel within postCall.agi, but even when 
the destination channel is definitely still up, the call returns an 
error "511 Command Not Permitted on a dead channel" (presumably because 
the current (caller) channel has hung up).


I can't find anything that I can use within the execution of the hangup 
extensions for the caller to determine whether the destination channel 
is still up.


Is it a bug that I only get the one call to the hangup extensions when 
both caller and callee channels hangup so close to each other that 
neither the "F" nor "g" flags have the desired effect?


Thanks in advance for any help.

John Hurst






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Re: [asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE

[Shaun Ruffell]

On Tue, Nov 20, 2012 at 05:18:12PM +0530, upendra wrote:
> Hi,
> 
> just going through the code i found that this  CONFIG_VOICEBUS_ECREFERENCE
> undef in the voicebus then how it will defined to run the echo cancell on
> the respective  drievers wctdm24xxp ??
>  explain how this  CONFIG_VOICEBUS_ECREFERENCE  enabled and where it is
> enabled while run time.

It's a compile time option that was in added to the code for testing
in certain environments [1]. It doesn't enable / disable the
echocanceler, but instead changes the default processing to provide
a more accurate reference signal when latency grows large.

[1] http://svnview.digium.com/svn/dahdi?view=revision&revision=9144

If you would like to use it for some reason, in
drivers/dahdi/voicebus.h change "#undef CONFIG_VOICEBUS_ECREFERENCE"
to "#define CONFIG_VOICEBUS_ECREFERENCE".

In practice, I've found that it's generally better to fix what on
the system is causing latency to grow so large instead of enabling
that option.

Cheers,
Shaun

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] leading ghost 0

[Danny Nicholas]

In my past experience the best recourse for dealing with a DAHDI trunked
asterisk system is this sequence

Service asterisk stop

Service dahdi restart

Service asterisk start

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Tuesday, November 20, 2012 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] leading ghost 0

 

Not only, you have to restart dahdi/zaptel as well.

 

Leandro

2012/11/20 Frederic Van Espen 

On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
> I'm sure nobody has added something... tried prilocaldialplan and
> pridialplan but nothing changed.
> Question: if pridialplan or prilocaldialplan would work, should I see
> the 0 inside PRI frame with intense debug or it is hidden?

Somebody correct me if I'm wrong but I think you have to restart
asterisk when you change these settings on dahdi. Keep that in mind.

Cheers,

Frederic



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Re: [asterisk-users] No more connections allowed.

[Danny Nicholas]

You've exceeded the allowed maximum number of simultaneous remote console
connections (128).  While it may be a bit aggressive to have it kill all
current connected consoles, its also a bit excessive to have
128 connected remote consoles.

While the behaviour may not be entirely desirable, this isn't so much a bug
as a limitation of the system.

--
Why would you need 128 simultaneous remote consoles?  IMO the number should
be something like 16 and a timeout option needs to be added so you don't
have unattended consoles consuming resources and opening holes.


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Re: [asterisk-users] leading ghost 0

[Leandro Dardini]

Not only, you have to restart dahdi/zaptel as well.

Leandro

2012/11/20 Frederic Van Espen 

> On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
> > I'm sure nobody has added something... tried prilocaldialplan and
> > pridialplan but nothing changed.
> > Question: if pridialplan or prilocaldialplan would work, should I see
> > the 0 inside PRI frame with intense debug or it is hidden?
>
> Somebody correct me if I'm wrong but I think you have to restart
> asterisk when you change these settings on dahdi. Keep that in mind.
>
> Cheers,
>
> Frederic
>
>
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Re: [asterisk-users] leading ghost 0

[Frederic Van Espen]

On Tue, 2012-11-20 at 15:03 +0100, gincantalupo wrote:
> I'm sure nobody has added something... tried prilocaldialplan and
> pridialplan but nothing changed.
> Question: if pridialplan or prilocaldialplan would work, should I see
> the 0 inside PRI frame with intense debug or it is hidden? 

Somebody correct me if I'm wrong but I think you have to restart
asterisk when you change these settings on dahdi. Keep that in mind.

Cheers,

Frederic


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Re: [asterisk-users] leading ghost 0

[gincantalupo]

Hi Leandro,

I'm sure nobody has added something... tried prilocaldialplan and 
pridialplan but nothing changed.
Question: if pridialplan or prilocaldialplan would work, should I see 
the 0 inside PRI frame with intense debug or it is hidden?


Yes...the technician did it...there is only one cable.

Maybe it is the socket circuitry that has something wrong but I do not 
know ho to check.


Asap I'll be on site I'll do more testing.

Thank you

Giorgio

On 11/20/2012 01:13 PM, Leandro Dardini wrote:
That is a real mistery! I like a lots these cases when all seems not 
working despite all being correctly configured, but you know first or 
later you'll find the answer.


From your website, it seems you are selling/renting PBX based on 
asterisk, so you can be sure nobody has messed with the asterisk or 
dahdi source code adding a zero... I am sure you have already tried 
with a brand new server.


Have you checked the pridialplan and prilocaldialplan setting?

If I was in your shoes, I'll get another server, with a PRI configured 
as master and hook it at your PBX to really check if the zero is sent.


Does the technician try to make phone calls from the same network 
cable you are using?


Leandro


2012/11/20 gincantalupo >


Hi Leandro,

thanks for your answer.

I already have tried those parameters but without any positive result.

The telco technician has tried the line with its machine and it
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle that adds
the zero but do not know what it is.
Strange is the fact that you can call some numbers with or without
the prefix zero...
Moreover we had no problem with the previous telco (fastweb).

So we can only call PTSN numbersnot mobile phones.

Giorgio


On 11/20/2012 11:12 AM, Leandro Dardini wrote:

2012/11/20 gincantalupo mailto:gincantal...@fgasoftware.com>>

Hi all,

I have problems dialling out because my new telco (the
previous gave no problems) tells me my PBX adds a leading 0
and that's why I cannot dial out (but I can receive calls).

I make a small extensions.conf as a test:

exten => 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten => 666,1,Dial(DAHDI/g1/0233xx)
and
exten => 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside frames

How can I really be SURE Asterisk is not adding some leading
zero?

Thank you.

Giorgio.


I have never heard of a way to automatically add digits when
using PRI, however can you check your chan_dahdi.conf about the
following lines:

internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in
Italy, every provider has PRI configured in its own way, some
time even the same provider is configuring PRI lines in multiple
times, but often the problems are on receiving the calls (like
calls with and without the area code, with or without the leading
zero, etc. etc.)

Leandro


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Re: [asterisk-users] No more connections allowed.

[Matthew Jordan]

On 11/20/2012 03:32 AM, Jonson Player wrote:
> Hello,
> 
> I have strange situation with asterisk 1.8.18.0 , randomly i got this
> message in cli:
> 
> "WARNING[15925] asterisk.c: No more connections allowed"
> 
> All connections freeze and all extensions doesn't work anymore. Is any
> bug or is any setting that can solve this problem?
> 
> Thank you.
> 

You've exceeded the allowed maximum number of simultaneous remote
console connections (128).  While it may be a bit aggressive to have it
kill all current connected consoles, its also a bit excessive to have
128 connected remote consoles.

While the behaviour may not be entirely desirable, this isn't so much a
bug as a limitation of the system.

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



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Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

[Joshua Colp]

Face wrote:


Well, thanks for responding. I went back to 10.10.0 and things seem to
be working fine now!


This is certainly good to know but I'd like to know why upgrading to 11 
did not seem to work for you. This is the first case since it's been out 
where it doesn't appear to have been smooth. Would you be willing to 
provide the information I asked about from a running 11 instance?


Cheers,

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Re: [asterisk-users] No more connections allowed.

[Joshua Colp]

Jonson Player wrote:

Hello,


Hola,


I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:

"WARNING[15925] asterisk.c: No more connections allowed"


This message is output when the number of Asterisk consoles (asterisk -r 
instances) has reached the limit. This limit is 128 by default which is 
quite a lot. Are you doing something that would cause a lot? Could they 
be hanging around by mistake?



All connections freeze and all extensions doesn't work anymore. Is any
bug or is any setting that can solve this problem?


This definitely shouldn't happen but it would be useful to know exactly 
what you are doing with the system. Answering my questions above is a 
good start.


Cheers,

--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] addressing peers dynamically

[Joshua Colp]

Andre Gronwald wrote:

it is just because i think that something is not wrong (which is
correct, because i address a currently not existing peer). and if there
is a way to handle it better, then i would like to know it (virtual
queues is just oversized, but maybe there is a simple usage of
addressing only registered peers...


Ultimately something has to check if the peer is available or not, the 
only difference is instead of having app_dial do it you do it to mask 
the message.


You could use ChanIsAvail on each peer you want to dial and use the 
result to construct a string of available devices.


Cheers,

--
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] leading ghost 0

[Leandro Dardini]

That is a real mistery! I like a lots these cases when all seems not
working despite all being correctly configured, but you know first or later
you'll find the answer.

>From your website, it seems you are selling/renting PBX based on asterisk,
so you can be sure nobody has messed with the asterisk or dahdi source code
adding a zero... I am sure you have already tried with a brand new server.

Have you checked the pridialplan and prilocaldialplan setting?

If I was in your shoes, I'll get another server, with a PRI configured as
master and hook it at your PBX to really check if the zero is sent.

Does the technician try to make phone calls from the same network cable you
are using?

Leandro


2012/11/20 gincantalupo 

> **
> Hi Leandro,
>
> thanks for your answer.
>
> I already have tried those parameters but without any positive result.
>
> The telco technician has tried the line with its machine and it
> worked...remote telco technicians say they get a leading zero...
> I'm thinking there is something strange in the middle that adds the zero
> but do not know what it is.
> Strange is the fact that you can call some numbers with or without the
> prefix zero...
> Moreover we had no problem with the previous telco (fastweb).
>
> So we can only call PTSN numbersnot mobile phones.
>
> Giorgio
>
>
> On 11/20/2012 11:12 AM, Leandro Dardini wrote:
>
> 2012/11/20 gincantalupo 
>
>> Hi all,
>>
>> I have problems dialling out because my new telco (the previous gave no
>> problems) tells me my PBX adds a leading 0 and that's why I cannot dial out
>> (but I can receive calls).
>>
>> I make a small extensions.conf as a test:
>>
>> exten => 666,1,Dial(DAHDI/g1/339xx)
>> but cannot dial out
>>
>> Curious thing is that
>> exten => 666,1,Dial(DAHDI/g1/0233xx)
>> and
>> exten => 666,1,Dial(DAHDI/g1/233xx)
>> call the same number!!!
>>
>> Line in use is a PRI.
>>
>> My Asterisk version is 1.4.26.2
>> dahdi version: 2.2.0.2
>> wanpipe-3.4.6
>>
>> I checked with intense pri debug and see no 0 inside frames
>>
>> How can I really be SURE Asterisk is not adding some leading zero?
>>
>> Thank you.
>>
>> Giorgio.
>>
>>
>  I have never heard of a way to automatically add digits when using PRI,
> however can you check your chan_dahdi.conf about the following lines:
>
>  internationalprefix =
> nationalprefix =
> localprefix =
>
>  If presents, try messing with them. If you are using the PRI in Italy,
> every provider has PRI configured in its own way, some time even the same
> provider is configuring PRI lines in multiple times, but often the problems
> are on receiving the calls (like calls with and without the area code, with
> or without the leading zero, etc. etc.)
>
>  Leandro
>
>
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>
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Re: [asterisk-users] leading ghost 0

[gincantalupo]

Hi Leandro,

thanks for your answer.

I already have tried those parameters but without any positive result.

The telco technician has tried the line with its machine and it 
worked...remote telco technicians say they get a leading zero...
I'm thinking there is something strange in the middle that adds the zero 
but do not know what it is.
Strange is the fact that you can call some numbers with or without the 
prefix zero...

Moreover we had no problem with the previous telco (fastweb).

So we can only call PTSN numbersnot mobile phones.

Giorgio

On 11/20/2012 11:12 AM, Leandro Dardini wrote:
2012/11/20 gincantalupo >


Hi all,

I have problems dialling out because my new telco (the previous
gave no problems) tells me my PBX adds a leading 0 and that's why
I cannot dial out (but I can receive calls).

I make a small extensions.conf as a test:

exten => 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten => 666,1,Dial(DAHDI/g1/0233xx)
and
exten => 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside frames

How can I really be SURE Asterisk is not adding some leading zero?

Thank you.

Giorgio.


I have never heard of a way to automatically add digits when using 
PRI, however can you check your chan_dahdi.conf about the following lines:


internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in Italy, 
every provider has PRI configured in its own way, some time even the 
same provider is configuring PRI lines in multiple times, but often 
the problems are on receiving the calls (like calls with and without 
the area code, with or without the leading zero, etc. etc.)


Leandro


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[asterisk-users] info on : CONFIG_VOICEBUS_ECREFERENCE

[upendra]

Hi,


just going through the code i found that this  CONFIG_VOICEBUS_ECREFERENCE
undef in the voicebus then how it will defined to run the echo cancell on
the respective  drievers wctdm24xxp ??
 explain how this  CONFIG_VOICEBUS_ECREFERENCE  enabled and where it is
enabled while run time.



regards
Uppi.
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[asterisk-users] 回覆︰ tcptls ssl connection error

[kingman chui]

Hi,
  I set up tls and srtp with lyn and asterisk 1.8 before.
I think your ssl connection is not setup . 
so, it may due to key and certificate problem.
If the key and cert is ok with CA, the ssl connection will up auto..
 
I work this before and I can connect to lync server with TLS and srtp 
I my case, lync server is CA auth , asterisk is the client only 
 
 
Hope this can help you ...
 

寄件人︰ Chandrakant Solanki 
>收件人︰ Asterisk Users Mailing List - Non-Commercial Discussion 
> 
>傳送日期︰ 2012年11月20日 (週二) 2:39 PM
>主題︰ Re: [asterisk-users] tcptls ssl connection error
>
>
>Hello All,
>
>Anyone have idea regarding below error.
>
>After applying all patch, still faced the same issue.
>
>
>--
>Regards,
>
>Chandrakant Solanki
>
>
>On Fri, Nov 9, 2012 at 11:39 AM, Chandrakant Solanki 
> wrote:
>>
>> Hello All,
>>
>> I am using asterisk 1.8.13.0 and which is running on TLS port and my request 
>> forwarded from opensips which is also run tls port.
>>
>> On both end my certificate is same.
>>
>> During search about this error, I found below blog and apply patch, then 
>> also found below error.
>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-18345
>> https://issues.asterisk.org/jira/browse/ASTERISK-20559
>> Also applied r375023
>>
>> [Nov  8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() 
>> failed: 5
>> [Nov  8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() 
>> failed: 5
>> [Nov  8 21:57:37]   == Problem setting up ssl connection: 
>> error::lib(0):func(0):reason(0)
>> [Nov  8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection: 
>> FILE * open failed!
>> [Nov  8 21:57:39]   == Problem setting up ssl connection: 
>> error::lib(0):func(0):reason(0)
>> [Nov  8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection: 
>> FILE * open failed!
>> [Nov  8 21:57:49]   == Problem setting up ssl connection: 
>> error::lib(0):func(0):reason(0)
>> [Nov  8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection: 
>> FILE * open failed!
>>
>>
>> --
>> Regards,
>>
>> Chandrakant Solanki
>
>
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Re: [asterisk-users] leading ghost 0

[Leandro Dardini]

2012/11/20 gincantalupo 

> Hi all,
>
> I have problems dialling out because my new telco (the previous gave no
> problems) tells me my PBX adds a leading 0 and that's why I cannot dial out
> (but I can receive calls).
>
> I make a small extensions.conf as a test:
>
> exten => 666,1,Dial(DAHDI/g1/339xx)
> but cannot dial out
>
> Curious thing is that
> exten => 666,1,Dial(DAHDI/g1/**0233xx)
> and
> exten => 666,1,Dial(DAHDI/g1/233xx)
> call the same number!!!
>
> Line in use is a PRI.
>
> My Asterisk version is 1.4.26.2
> dahdi version: 2.2.0.2
> wanpipe-3.4.6
>
> I checked with intense pri debug and see no 0 inside frames
>
> How can I really be SURE Asterisk is not adding some leading zero?
>
> Thank you.
>
> Giorgio.
>
>
I have never heard of a way to automatically add digits when using PRI,
however can you check your chan_dahdi.conf about the following lines:

internationalprefix =
nationalprefix =
localprefix =

If presents, try messing with them. If you are using the PRI in Italy,
every provider has PRI configured in its own way, some time even the same
provider is configuring PRI lines in multiple times, but often the problems
are on receiving the calls (like calls with and without the area code, with
or without the leading zero, etc. etc.)

Leandro
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[asterisk-users] leading ghost 0

[gincantalupo]

Hi all,

I have problems dialling out because my new telco (the previous gave no 
problems) tells me my PBX adds a leading 0 and that's why I cannot dial 
out (but I can receive calls).


I make a small extensions.conf as a test:

exten => 666,1,Dial(DAHDI/g1/339xx)
but cannot dial out

Curious thing is that
exten => 666,1,Dial(DAHDI/g1/0233xx)
and
exten => 666,1,Dial(DAHDI/g1/233xx)
call the same number!!!

Line in use is a PRI.

My Asterisk version is 1.4.26.2
dahdi version: 2.2.0.2
wanpipe-3.4.6

I checked with intense pri debug and see no 0 inside frames

How can I really be SURE Asterisk is not adding some leading zero?

Thank you.

Giorgio.

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[asterisk-users] No more connections allowed.

[Jonson Player]

Hello,

I have strange situation with asterisk 1.8.18.0 , randomly i got this
message in cli:

"WARNING[15925] asterisk.c: No more connections allowed"

All connections freeze and all extensions doesn't work anymore. Is any bug
or is any setting that can solve this problem?

Thank you.

Jonson.
http://mobile-sip.tel/
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