Re: [asterisk-users] Fwd: Backport fix

2016-08-17 Thread Joshua Colp 

Saint Michael wrote:


There is a big bug
https://issues.asterisk.org/jira/browse/ASTERISK-24768

It affects version 11, but it was fixed only from 13.20 onwards.
However, millions of people still use version 11. This bug makes
Asterisk crash every few hours under any load that has RTP going through
Asterisk. For example, after
3 hours and 6 minutes
  lsof | grep asterisk | grep FIFO | wc -l
245122
with less than 120 calls, with fill media proxy.
and it crashes
Can somebody help?


The specific issue you are referring to is not applicable to 11, the 
issue was a regression only applicable to 13 and above. Per your query 
on the issue itself, the fix was not placed into certified 13.1 as no 
Digium customer requested it to be.


--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org


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[asterisk-users] Fwd: Backport fix

2016-08-17 Thread Saint Michael 
There is a big bug
https://issues.asterisk.org/jira/browse/ASTERISK-24768
It affects version 11, but it was fixed only from 13.20 onwards.
However, millions of people still use version 11. This bug makes Asterisk
crash every few hours under any load that has RTP going through Asterisk.
For example, after
3 hours and 6 minutes
 lsof | grep asterisk | grep FIFO | wc -l
245122
with less than 120 calls, with fill media proxy.
and it crashes
Can somebody help?
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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-17 Thread George Joseph 
On Wed, Aug 17, 2016 at 1:40 PM, Jonas Kellens 
wrote:

> On 16-08-16 17:45, George Joseph wrote:
>
>
>
> On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens 
> wrote:
>
>> On 16-08-16 04:38, George Joseph wrote:
>>
>>
>>
>> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens 
>> wrote:
>>
>>> Hello
>>>
>>> using pjproject 2.5.5
>>> using asterisk-certified-13.8-cert1
>>>
>>
>> IIRC there were API changes in pjproject 2.5 that aren't accounted for in
>> asterisk 13.8.  Try pjproject 2.4.5 first and let's see if that works
>>
>>
>>>
>>> Compiled pjproject 2.5.5 with :
>>> ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
>>> --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound
>>> --disable-opencore-amr
>>>
>>> Compiled Asterisk 13 with
>>> ./configure --libdir=/usr/lib64
>>>
>>> All pjproject modules are selectable in menuselect, so here no problem.
>>>
>>> Modules are present in /usr/lib64/asterisk/module (see below).
>>>
>>> But when I start asterisk, I get a lot of errors concerning res_pjsip
>>> (see below) on the asterisk CLI.
>>>
>>> Anyone have some input on this ?
>>>
>>>
>>> Thanks.
>>>
>>> Kind regards.
>>>
>>>
>>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
>>
>> Hello
>>
>> how can I disable all modules related to pjsip in modules.conf ??
>>
>> I have now :
>>
>> [modules]
>> autoload=yes
>> preload => res_config_mysql.so
>> noload => pbx_gtkconsole.so
>> noload => res_pjsip.so
>> noload => res_pjsip_pubsub.so
>> noload => res_pjsip_session.so
>> noload => chan_pjsip.so
>> noload => res_pjsip_exten_state.so
>> noload => res_pjsip_log_forwarder.so
>> load => res_musiconhold.so
>> noload => chan_alsa.so
>> noload => chan_oss.so
>> noload => chan_console.so
>>
>>
>> This does not make the CLI erros go away. I still have the idea that
>> pjsip is loaded.
>>
>>
>
> I'm not sure what your objective is.  If you want to completely disable
> pjsip, run ./configure --without-pjproject.
>
>
> When I compile "--without-pjproject" I loose all webrtc functionality. I
> get errors about the lack of "ice-frag and ice-pwd in the SDP-body".
>
> So I guess I DO need pjproject. But I do not want to use pjsip (I prefer
> sip).
>
> Do you have any other input or idea ?
>

Ok, I get it now.  Use pjproject-2.4.5 and in menuselect, disable all the
res_pjsip modules.


>
>
> Kind regards.
>
> J.
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>



-- 
George Joseph
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445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-17 Thread Jonathan H 
On 17 August 2016 at 20:40, Jonas Kellens  wrote:

> When I compile "--without-pjproject" I loose all webrtc functionality. I get 
> errors about the lack of "ice-frag and ice-pwd in the SDP-body".
> So I guess I DO need pjproject. But I do not want to use pjsip (I prefer sip).
> Do you have any other input or idea ?

Yes.

I've never had a problem compiling or installing Asterisk; I simply
download the latest version, follow the instructions, and 10 minutes
later I'm compiled and up and running.
No messing about with weird seperate downloads of unsupported versions
of pjsip - I just use the bundled pjsip install and off I go.

But from your posts, it seems you want to do modern web stuff like
WebRTC and so on, on old version of centos, old versions of asterisk,
old version of the SIP channel driver.

What particular reason is there to even bother with the certified
version - the instructions say the regular most recent LTS download
should be first choice.

And why do you prefer SIP? pjsip was introduced in Asterisk 12 nearly
3 years ago, and SIP is pretty much deprecated now.

As a newbie, I looked at SIP and it all seemed a bit bonkers -
"type=friend, insecure=very" - what's THAT all about?!

In pjsip, I just setup a pjsip_wizard and template my endpoints in
pjsip.conf, and I'm done in a few lines.
https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md

This is me, creating a brand new Asterisk install on a low end $5 VPS
which handles more concurrent calls than I need it to (at least 20 so
far!);
https://www.youtube.com/watch?v=h12NkJQwpYo (I just found out that the
Youtube annotations don't work on mobile, so watch on desktop for it
to make sense!).

I'm probably the newbiest of noobs here, but just using the latest
current stable version of everything available and following the
install page on the Asterisk Wiki I can fire up a VPS and be receiving
calls in 20 minutes, from scratch. And I'm genuinely interested in why
people struggle on for days with old versions of things. I'm not
asking all this to create argument, but I am genuinely interested.
Perhaps I am missing a major point here?

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Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Matthew Jordan 
On Wed, Aug 17, 2016 at 8:29 AM, Hooman Fazaeli  wrote:
> On 2016-08-16 12:10, Joris Engbers wrote:
>>
>> Hooman Fazaeli writes:
>>
>>> Hi
>>>
>>> I have noticed that asterisk returns 'SIP 603' when the called party does
>>> not answer.
>>>
>>> My test setup is simple: two SIP phones (extensions: 100 and 111)
>>> registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30
>>> seconds.
>>> When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request
>>> to
>>> 111 (expected) and a '603 Decline' response to 100 (unexpected &
>>> misleading).
>>> It seems to me that a'480 Temporarily unavailable' response is more
>>> suitable in this situation.
>>>
>>> Is this a normal behavior of asterisk or a bug?
>>>
>>> Thanks.
>>
>> That sounds like you are not doing a Hangup().
>>
>> What is the dialplan that you are using?
>>
>
> Hangup() is there. The dial plan is:
>
> (I set dial timeout to 10s to speed up tests)
>
> [phone-100]
> exten => 111,1,Dial(SIP/111,10,tTo)
> exten => 111,n,Hangup()
>
> [phone-111]
> exten => 100,1,Dial(SIP/100,10,tTo)
> exten => 100,n,Hangup()
>
> As can be seen in below log messages, asterisk correctly sets DIALSTATUS to
> NOANSWER (line 7).
> Line 18 shows that the hangupcause value has been set to 16
> (AST_CAUSE_NORMAL_CLEARING) which
> asterisk complains has no SIP equivalent and falls back to 603. The problem
> seems to be
> that hangupcause is set incorrectly in the first place.
>
> ...
> 
> ...
> 1 VERBOSE[-1]: app_dial.c:1633 in wait_for_answer: -- Nobody picked up
> in 1 ms
> 2 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
> 'SIP/111-0003'
> 3 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/111-0003,
> SIP callid 1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060
> 4 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter
> for outgoing call
> 5 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state
> Ringing (not UP)
> ...
> 6 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'CANCEL sip:' onto
> UDP socket destined for 192.168.1.200:5062
> 7 DEBUG[-1]: app_dial.c:3033 in dial_exec_full: Exiting with
> DIALSTATUS=NOANSWER.
> 8 DEBUG[-1]: pbx.c:4720 in pbx_extension_helper: Launching 'Hangup'
> 9 VERBOSE[-1]: pbx.c:4728 in pbx_extension_helper: -- Executing
> [111@phone-100:2] Hangup("SIP/100-0002", "") in new stack
> 10 DEBUG[-1]: pbx.c:5544 in __ast_pbx_run: Spawn extension (phone-100,111,2)
> exited non-zero on 'SIP/100-0002'
> 11 VERBOSE[-1]: pbx.c:5545 in __ast_pbx_run:   == Spawn extension
> (SIP-PHONE-35145790056fd369709fb2, 111, 2) exited non-zero on
> 'SIP/100-0002'
> 12 DEBUG[-1]: channel.c:2735 in ast_softhangup_nolock: Soft-Hanging up
> channel 'SIP/100-0002'
> 13 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel
> 'SIP/100-0002'
> 14 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/100-0002,
> SIP callid 9eda334cf9584d408ccd6e14eae7143a
> 15 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter
> for incoming call
> 16 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state
> Ring (not UP)
> 17 DEBUG[-1]: res_rtp_asterisk.c:2604 in ast_rtp_remote_address_set: Setting
> RTCP address on RTP instance '0x29dbf01c'
> 18 DEBUG[-1]: chan_sip.c:6484 in hangup_cause2sip: AST hangup cause 16 (no
> match found in SIP)
> 19 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'SIP/2.0 603'
> onto UDP socket destined for 192.168.1.30:52628
> 20 ...
> 21 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
> checking channel drivers for SIP - 111
> 22 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 111
> 23 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/111 - state 1 (Not in use)
> 24 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state
> '1'
> 25 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
> checking channel drivers for SIP - 111
> 26 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 111
> 27 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/111 - state 1 (Not in use)
> 28 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state
> '1'
> 29 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found,
> checking channel drivers for SIP - 100
> 30 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for
> peer 100
> 31 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for
> SIP/100 - state 5 (Unavailable)
> 32 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100' state
> '5'
> 33 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID:
> 9eda334cf9584d408ccd6e14eae7143a (Checking From) --From tag
> 48505f8775334f429a54d48d5b095543 --To-tag as2ad3ad05
> 34 DEBUG[-1]: chan_sip.c:25954 in handle_incoming:  Received ACK

Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

2016-08-17 Thread Jonas Kellens 

On 16-08-16 17:45, George Joseph wrote:



On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens 
> wrote:


On 16-08-16 04:38, George Joseph wrote:



On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
> wrote:

Hello

using pjproject 2.5.5
using asterisk-certified-13.8-cert1


IIRC there were API changes in pjproject 2.5 that aren't
accounted for in asterisk 13.8.  Try pjproject 2.4.5 first and
let's see if that works


Compiled pjproject 2.5.5 with :
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr
--libdir=/usr/lib64 --enable-shared --disable-video
--disable-sound --disable-opencore-amr

Compiled Asterisk 13 with
./configure --libdir=/usr/lib64

All pjproject modules are selectable in menuselect, so here
no problem.

Modules are present in /usr/lib64/asterisk/module (see below).

But when I start asterisk, I get a lot of errors concerning
res_pjsip (see below) on the asterisk CLI.

Anyone have some input on this ?


Thanks.

Kind regards.





-- 
George Joseph

Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com  &
www.asterisk.org 



Hello

how can I disable all modules related to pjsip in modules.conf ??

I have now :

[modules]
autoload=yes
preload => res_config_mysql.so
noload => pbx_gtkconsole.so
noload => res_pjsip.so
noload => res_pjsip_pubsub.so
noload => res_pjsip_session.so
noload => chan_pjsip.so
noload => res_pjsip_exten_state.so
noload => res_pjsip_log_forwarder.so
load => res_musiconhold.so
noload => chan_alsa.so
noload => chan_oss.so
noload => chan_console.so


This does not make the CLI erros go away. I still have the idea
that pjsip is loaded.



I'm not sure what your objective is.  If you want to completely 
disable pjsip, run ./configure --without-pjproject.


When I compile "--without-pjproject" I loose all webrtc functionality. I 
get errors about the lack of "ice-frag and ice-pwd in the SDP-body".


So I guess I DO need pjproject. But I do not want to use pjsip (I prefer 
sip).


Do you have any other input or idea ?


Kind regards.

J.

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[asterisk-users] SIP client open URL upon answer (was : Re: SIP client able to handle Access-URL: header)

2016-08-17 Thread Bertrand LUPART - Linkeo.com 
Hello,


> I'm playing with the optional URL parameter of the Dial() command, which 
> "will also be sent to the called party upon successful connection, if the 
> channel technology supports the sending of URLs in this way."[1]
> 
> Basically, the following asterisk dialplan directive :
> 
> - - 8< - - 8< - - 8< - -
> same => n,Dial(SIP/1234,30,tr,http://www.example.com/)
> - - >8 - - >8 - - >8 - -
> 
> Will generate the following SIP header in the INVITE request :
> 
> - - 8< - - 8< - - 8< - -
> Access-URL: ;mode=active
> - - >8 - - >8 - - >8 - -
> 
> Does anyone know about a SIP client able to handle this header ? I'd expect 
> it to automatically open this webpage.

Self-answering after some research on this topic.

Seems like "Access-URL" isn't an official SIP header :
http://www.iana.org/assignments/sip-parameters/sip-parameters.xhtml#sip-parameters-2

This Dial() optional URL parameter looks like the same beast than SendURL() 
added nearly 10 years ago in asterisk SIP stack :
https://github.com/asterisk/asterisk/commit/04e45cfda35bd3cb4e508aad9286b5deb99ccf56

Is anyone aware of a SIP client which would open an URL sent by asterisk?


Thank you,

-- 
Bertrand LUPART

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Re: [asterisk-users] Farewell

2016-08-17 Thread Josh Reynolds 
Congrats! Enjoy the time away, you've earned it :)

---
Josh Reynolds
josh@engineered.online

On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina  wrote:

> I just wanted to wish all of you good luck I'm officially retired and will
> be removing my name from the list. I can attest that this list has been a
> great help throughout my career. I have deployed probably over 100
> installations over a 10-year period.
>
> Any of you newcomers this list the most valuable tool you can have.
>
>
>
> Sincerely,
>
> Vincent Medina
> Information Systems Director
> APCN, Inc.
>
> (305)785-3355
>
> Sent using www.apcn.net Internet Services.
>
>
>  Original message 
> From: Dario Estupinan 
> Date: 08/17/2016 8:53 AM (GMT-05:00)
> To: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Subject: Re: [asterisk-users] Realtime SIP peers do not register any more
> after upgrade to Asterisk 13
>
> REMOVE ME please.
>
> 2016-08-15 15:16 GMT-05:00 Jonas Kellens :
>
>> Hello
>>
>> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
>> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>>
>>
>> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5076' - Wrong password
>> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5072' - Wrong password
>> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5062' - Wrong password
>> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
>> handle_request_register: Registration from ''
>> failed for '78.119.140.190:5060' - Wrong password
>>
>>
>> Is this a known problem ??
>>
>>
>> Second question I have : can I get the complete list of columns that can
>> be used in realtime database for sip peers somewhere (update for Ast 13) ?
>> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
>> dtlssetup possible ??
>>
>>
>>
>>
>> Thanks for the help.
>>
>>
>> Kind regards.
>>
>> Jonas.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
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[asterisk-users] Farewell

2016-08-17 Thread Vincent Medina 


I just wanted to wish all of you good luck I'm officially retired and will be 
removing my name from the list. I can attest that this list has been a great 
help throughout my career. I have deployed probably over 100 installations over 
a 10-year period. 
Any of you newcomers this list the most valuable tool you can have. 


Sincerely, 
Vincent MedinaInformation Systems DirectorAPCN, Inc.
(305)785-3355
Sent using www.apcn.net Internet Services.

 Original message 
From: Dario Estupinan  
Date: 08/17/2016  8:53 AM  (GMT-05:00) 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
 
Subject: Re: [asterisk-users] Realtime SIP peers do not register any more after 
upgrade to Asterisk 13 

REMOVE ME please.
2016-08-15 15:16 GMT-05:00 Jonas Kellens :
Hello



after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1 none of 
my realtime SIP peers (saved in MySQL DB) register anymore.





[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: 
Registration from '' failed for 
'78.119.140.190:5076' - Wrong password

[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register: 
Registration from '' failed for 
'78.119.140.190:5072' - Wrong password

[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register: 
Registration from '' failed for 
'78.119.140.190:5062' - Wrong password

[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register: 
Registration from '' failed for 
'78.119.140.190:5060' - Wrong password

[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register: 
Registration from '' failed for 
'78.119.140.190:5060' - Wrong password





Is this a known problem ??





Second question I have : can I get the complete list of columns that can be 
used in realtime database for sip peers somewhere (update for Ast 13) ? Are 
columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup 
possible ??









Thanks for the help.





Kind regards.



Jonas.



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Re: [asterisk-users] SIP 603 response when call is not answered

2016-08-17 Thread Hooman Fazaeli 

On 2016-08-16 12:10, Joris Engbers wrote:

Hooman Fazaeli writes:


Hi

I have noticed that asterisk returns 'SIP 603' when the called party does
not answer.

My test setup is simple: two SIP phones (extensions: 100 and 111)
registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds.
When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to
111 (expected) and a '603 Decline' response to 100 (unexpected &
misleading).
It seems to me that a'480 Temporarily unavailable' response is more
suitable in this situation.

Is this a normal behavior of asterisk or a bug?

Thanks.

That sounds like you are not doing a Hangup().

What is the dialplan that you are using?



Hangup() is there. The dial plan is:

(I set dial timeout to 10s to speed up tests)

[phone-100]
exten => 111,1,Dial(SIP/111,10,tTo)
exten => 111,n,Hangup()

[phone-111]
exten => 100,1,Dial(SIP/100,10,tTo)
exten => 100,n,Hangup()

As can be seen in below log messages, asterisk correctly sets DIALSTATUS to 
NOANSWER (line 7).
Line 18 shows that the hangupcause value has been set to 16 
(AST_CAUSE_NORMAL_CLEARING) which
asterisk complains has no SIP equivalent and falls back to 603. The problem 
seems to be
that hangupcause is set incorrectly in the first place.

...

...
1 VERBOSE[-1]: app_dial.c:1633 in wait_for_answer: -- Nobody picked up in 
1 ms
2 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel 'SIP/111-0003'
3 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/111-0003, SIP 
callid 1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060
4 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter for 
outgoing call
5 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state Ringing 
(not UP)
...
6 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'CANCEL sip:' onto 
UDP socket destined for 192.168.1.200:5062
7 DEBUG[-1]: app_dial.c:3033 in dial_exec_full: Exiting with 
DIALSTATUS=NOANSWER.
8 DEBUG[-1]: pbx.c:4720 in pbx_extension_helper: Launching 'Hangup'
9 VERBOSE[-1]: pbx.c:4728 in pbx_extension_helper: -- Executing [111@phone-100:2] 
Hangup("SIP/100-0002", "") in new stack
10 DEBUG[-1]: pbx.c:5544 in __ast_pbx_run: Spawn extension (phone-100,111,2) 
exited non-zero on 'SIP/100-0002'
11 VERBOSE[-1]: pbx.c:5545 in __ast_pbx_run:   == Spawn extension 
(SIP-PHONE-35145790056fd369709fb2, 111, 2) exited non-zero on 'SIP/100-0002'
12 DEBUG[-1]: channel.c:2735 in ast_softhangup_nolock: Soft-Hanging up channel 
'SIP/100-0002'
13 DEBUG[-1]: channel.c:2884 in ast_hangup: Hanging up channel 
'SIP/100-0002'
14 DEBUG[-1]: chan_sip.c:6553 in sip_hangup: Hangup call SIP/100-0002, SIP 
callid 9eda334cf9584d408ccd6e14eae7143a
15 DEBUG[-1]: chan_sip.c:6169 in update_call_counter: Updating call counter for 
incoming call
16 DEBUG[-1]: chan_sip.c:6572 in sip_hangup: Hanging up channel in state Ring 
(not UP)
17 DEBUG[-1]: res_rtp_asterisk.c:2604 in ast_rtp_remote_address_set: Setting 
RTCP address on RTP instance '0x29dbf01c'
18 DEBUG[-1]: chan_sip.c:6484 in hangup_cause2sip: AST hangup cause 16 (no 
match found in SIP)
19 DEBUG[-1]: chan_sip.c:3554 in __sip_xmit: Trying to put 'SIP/2.0 603' onto 
UDP socket destined for 192.168.1.30:52628
20 ...
21 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found, 
checking channel drivers for SIP - 111
22 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for 
peer 111
23 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/111 
- state 1 (Not in use)
24 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state '1'
25 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found, 
checking channel drivers for SIP - 111
26 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for 
peer 111
27 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/111 
- state 1 (Not in use)
28 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/111' state '1'
29 DEBUG[-1]: devicestate.c:344 in _ast_device_state: No provider found, 
checking channel drivers for SIP - 100
30 DEBUG[-1]: chan_sip.c:27264 in sip_devicestate: Checking device state for 
peer 100
31 DEBUG[-1]: devicestate.c:467 in do_state_change: Changing state for SIP/100 
- state 5 (Unavailable)
32 DEBUG[-1]: devicestate.c:442 in devstate_event: device 'SIP/100' state '5'
33 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID: 
9eda334cf9584d408ccd6e14eae7143a (Checking From) --From tag 
48505f8775334f429a54d48d5b095543 --To-tag as2ad3ad05
34 DEBUG[-1]: chan_sip.c:25954 in handle_incoming:  Received ACK (6) - 
Command in SIP ACK
35 DEBUG[-1]: chan_sip.c:4238 in __sip_ack: Stopping retransmission on 
'9eda334cf9584d408ccd6e14eae7143a' of Response 21834: Match Found
36 DEBUG[-1]: chan_sip.c:8436 in find_call: = Looking for  Call ID: 
1a8ef4ce3f4d8a513de4639916c28b15@192.168.1.17:5060 (Checking To) --From tag

Re: [asterisk-users] How to remove unused custom hints?

2016-08-17 Thread Tomáš Holý 
Perfect. That is exactly what I need.

Thank you very much.

Nice day!

S pozdravem

Tomáš Holý
INTERCONNECT s.r.o.
Zákaznická linka: +420 61333
TEL: +420 61321
FAX: +420 246063179
h...@interconnect.cz


Dne úterý 16. srpna 2016 12:01:20 CEST, John Kiniston napsal(a):


You can delete them from the astdb with database del.



do a 'database show' and your devices should show up in the tree under 
'CustomDevstate'




On Mon, Aug 15, 2016 at 9:42 AM, Tomas Holy  wrote:


Hello list members,
after programing of dialplan I have some messy Custom:hints which I can see in 
'devstate list'. I didn't find any possibility how to remove this hints from 
Asterisk and I 
want remove them. 


Can you help me with that, please? I tried search about that something in 
documentation or on Google, but I didn't find anything. 


asterisk*CLI> devstate list  

- --- 
Custom Device States 
 
 --- 
Name: 
'Custom:queuememberCALLERID(num)'  State: 'RINGING' --- --- Name: 
'Custom:queuememberh'  State: 'NOT_INUSE' --- --- Name: 'Custom:queuemembers'  
State: 'INUSE' ---



Thank you


Have a nice day!




S pozdravem

Tomáš HolýINTERCONNECT s.r.o.Zákaznická linka: +420 61333TEL: +420 
61321FAX: +420 246063179

_holy@interconnect.cz_

http://www.api-digital.com[2] --New to Asterisk? Join us for a live 
introductory webinar 
every Thurs:   http://www.asterisk.org/hello[3]
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hog, 
conn a ship, design a building, write a sonnet, balance accounts, build a wall, 
set a 
bone, comfort the dying, take orders, give orders, cooperate, act alone, solve 
equations, analyze a new problem, pitch manure, program a computer, cook a 
tasty 
meal, fight efficiently, die gallantly. Specialization is for 
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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread Jonas Kellens 

Remove yourself !

Don't hijack my thread !



On 17-08-16 14:53, Dario Estupinan wrote:

REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens >:


Hello

after I have upgraded from Asterisk 12 to
asterisk-certified-13.8-cert1 none of my realtime SIP peers (saved
in MySQL DB) register anymore.


[Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5076 ' - Wrong password
[Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5072 ' - Wrong password
[Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5062 ' - Wrong password
[Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password
[Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451
handle_request_register: Registration from
'>' failed for
'78.119.140.190:5060 ' - Wrong password


Is this a known problem ??


Second question I have : can I get the complete list of columns
that can be used in realtime database for sip peers somewhere
(update for Ast 13) ? Are columns like dtlsenable, dtlsverify,
dtlscertfile, dtlscafile, dtlssetup possible ??




Thanks for the help.


Kind regards.

Jonas.

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AVISO LEGAL: Este mensaje es confidencial, puede contener información 
privilegiada y no puede ser usado ni divulgado por personas distintas 
de su destinatario. Si recibe este correo por error, por favor 
elimínelo y avise a su remitente. Está prohibida su retención, 
grabación, utilización, aprovechamiento o divulgación con cualquier 
propósito. La Corporación Politécnica Nacional de Colombia no asume 
ninguna responsabilidad por eventuales daños generados por el recibo y 
el uso de este material, siendo responsabilidad del destinatario 
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defectos. El presente correo electrónico solo refleja la opinión de su 
Remitente y no representa necesariamente la opinión oficial de la 
Corporación o de sus Directivos.






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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread John Novack 

Remove yourself

READ - Included with every message -

asterisk-users mailing list
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Dario Estupinan wrote:


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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread Dario Estupinan 
REMOVE ME please.

2016-08-15 15:16 GMT-05:00 Jonas Kellens :

> Hello
>
> after I have upgraded from Asterisk 12 to asterisk-certified-13.8-cert1
> none of my realtime SIP peers (saved in MySQL DB) register anymore.
>
>
> [Aug 15 22:03:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5076' - Wrong password
> [Aug 15 22:04:13] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5072' - Wrong password
> [Aug 15 22:04:43] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5062' - Wrong password
> [Aug 15 22:04:46] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
> [Aug 15 22:04:53] NOTICE[30098]: chan_sip.c:28451 handle_request_register:
> Registration from '' failed for '
> 78.119.140.190:5060' - Wrong password
>
>
> Is this a known problem ??
>
>
> Second question I have : can I get the complete list of columns that can
> be used in realtime database for sip peers somewhere (update for Ast 13) ?
> Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile,
> dtlssetup possible ??
>
>
>
>
> Thanks for the help.
>
>
> Kind regards.
>
> 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
>



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Politécnica Nacional de Colombia no asume ninguna responsabilidad por
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Re: [asterisk-users] Realtime SIP peers do not register any more after upgrade to Asterisk 13

2016-08-17 Thread Jonas Kellens 


On 15-08-16 23:00, Carlos Chavez wrote:



I highly recommend that you use alembic to set up your database as 
this will make sure you are always using the correct database schema.  
You should be able to find the "official" structure in the 
contrib/realtime/mysql directory of the Asterisk source.




Hello

in contrib/realtime/mysql I see a table 'sippeers' with a column 
"transport ENUM('udp','tcp','tls','ws','wss','udp,tcp','tcp,udp') " but 
I see no columns dtlsenable, dtlsverify, dtlscertfile, dtlscafile, 
dtlssetup ?


So if we can define a sip peer with transport 'ws' or 'wss', then why 
are there no columns for the 'dtls'-part (which is kinda mandatory) ?




Kind regards.



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