[asterisk-users] WSS over Asterisk

[Steve Ng]

Hi,

Have anyone tried using SIPML5 to connect to Asterisk over wss?

I'm having the error as shown below

Connecting to 'wss://54.xxx.xxx.xxx:8080/ws '
SIPml-api.js?svn=224:1
==stack event = starting SIPml-api.js?svn=224:1
__tsip_transport_ws_onerror SIPml-api.js?svn=224:1
__tsip_transport_ws_onclose SIPml-api.js?svn=224:1
==stack event = failed_to_start


Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works
fine. Any idea why?
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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Tiago Geada]

Hi,

Let me append some extra info

cdr variable foo, shows on database, but value 'bar' doens't

its not even shown in the insert query

I tried with master_channel but no change


On 10 June 2014 16:25, Eric Wieling  wrote:

> Using Set(MASTER_CHANNEL(CDR(remoteUid))=foo); might do what you want
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mikael Fredin
> *Sent:* Tuesday, June 10, 2014 11:18 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] CDR custom variable on second call leg -
> via originate or .call file
>
>
>
> As far as I know, only way to set variables on another channel would be:
>
>  asterisk -rx "core show help dialplan set chanvar"
> Usage: dialplan set chanvar   
>Set channel variable  to 
>
>
>
>
>
> On 10 June 2014 16:39, Tiago Geada  wrote:
>
> Hi
>
>
>
>
>
> We have the following test .call file and test dialplan:
>
>
>
> I can't set a custom CDR var to a value on one channel leg, and another
> value on the connected channel leg?
>
>
>
>
>
> Is there a way I can woraround this issue?
>
>
>
>
>
>
>
> ## test call file
>
>
>
> Channel: Local/queue@TiagoGeada
>
> CallerID: teste-geada:0:210332450:
>
> MaxRetries: 0
>
> RetryTime: 1
>
> WaitTime: 8640
>
> Account: teste-geada
>
> Context: TiagoGeada
>
> Extension: outbound
>
> Archive: Yes
>
>
>
>
>
>
>
>
>
> ## dialplan
>
>
>
> queue => {
>
> Set(CDR(remoteUid)=foo);
>
> Queue(TiagoGeada,t,,,100);
>
> Hangup();
>
> }
>
>
>
> outbound => {
>
> //NoCDR();
>
> //ForkCDR(vdD);
>
> //ResetCDR(v);
>
> Set(CDR(remoteUid,r)=bar);
>
> Dial(Local/932485457@outbound,,gT);
>
> Hangup();
>
> }
>
>
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Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing

[Tzafrir Cohen]

On Tue, Jun 03, 2014 at 10:26:26PM +0200, Bart Remmerie wrote:
> Does anybody know where imap_tk is supposed to be / where it comes from ?
> Is it a part of asterisk / imap / linux / ...
> 
> I can't seem to find any references other than related to asterisk, but in 
> asterisk I only can find it as a (unfortunately missing) dependency for imap 
> support for voicemail...

What distribution is it?

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[asterisk-users] Hold

[Kelly Opal]

Hi
I am trying to set up a hold system so that a call is always parked in 
the same spot no matter how many times it is picked up. My problem is I cannot 
fins a variable the identifies the call all the way through until it is 
destroyed. ${UNIQUEID} and ${CHANNEL}  both seam to get lost when the call is 
parked. I tried setting 
set($[“${UNIQUEID}-hold”=”701”])
and
set($[“${CHANNEL}-hold”=”701”])

and both work fine until I do a transfer to park. Then both variables are 
blank. Is there any variable that is persistent to a call through all of the 
transfers.

asterisk 11.6-cert1
centos 5.7

Thanks

Kelly
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Re: [asterisk-users] Hold

[jonathan white]

Can you write the unique variable to astdb and then write it back to the
variable?

Not sure I have thought this through

J
On 11 Jun 2014 18:42, "Kelly Opal"  wrote:

>   Hi
> I am trying to set up a hold system so that a call is always
> parked in the same spot no matter how many times it is picked up. My
> problem is I cannot fins a variable the identifies the call all the way
> through until it is destroyed. ${UNIQUEID} and ${CHANNEL}  both seam to get
> lost when the call is parked. I tried setting
> set($[“${UNIQUEID}-hold”=”701”])
> and
> set($[“${CHANNEL}-hold”=”701”])
>
> and both work fine until I do a transfer to park. Then both variables are
> blank. Is there any variable that is persistent to a call through all of
> the transfers.
>
> asterisk 11.6-cert1
> centos 5.7
>
> Thanks
>
> Kelly
>
>
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Re: [asterisk-users] Hold

[Matthew Jordan]

On Wed, Jun 11, 2014 at 12:45 PM, jonathan white  wrote:

> Can you write the unique variable to astdb and then write it back to the
> variable?
>
> Not sure I have thought this through
>
> J
> On 11 Jun 2014 18:42, "Kelly Opal"  wrote:
>
>>   Hi
>> I am trying to set up a hold system so that a call is always
>> parked in the same spot no matter how many times it is picked up. My
>> problem is I cannot fins a variable the identifies the call all the way
>> through until it is destroyed. ${UNIQUEID} and ${CHANNEL}  both seam to get
>> lost when the call is parked. I tried setting
>> set($[“${UNIQUEID}-hold”=”701”])
>> and
>> set($[“${CHANNEL}-hold”=”701”])
>>
>> and both work fine until I do a transfer to park. Then both variables are
>> blank. Is there any variable that is persistent to a call through all of
>> the transfers.
>>
>> asterisk 11.6-cert1
>> centos 5.7
>>
>> Thanks
>>
>> Kelly
>>
>>
You are most likely running into masquerades. A masquerade is an internal
operation in Asterisk that involves renaming a channel. When this occurs,
your AMI client will receive a sequence of Masquerade and Rename events.
Your client will need to update its tracking of the channel based on those
events.

Alternatively, you can move to Asterisk 12. One of the major projects that
was done in that version was to remove the visibility of masquerades from
external systems (and mostly purge them internally), such that channels
have a stable, consistent identifier for the channel throughout its
lifetime.

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

[Matthew Jordan]

On Wed, Jun 11, 2014 at 9:10 AM, Tiago Geada  wrote:

> Hi,
>
> Let me append some extra info
>
> cdr variable foo, shows on database, but value 'bar' doens't
>
> its not even shown in the insert query
>
> I tried with master_channel but no change
>
>
I think you need to be a bit more specific about what CDR records you're
getting and what you'd like to have happen.

You have the following call file:




>
>>
>>
>>
>> ## test call file
>>
>>
>>
>> Channel: Local/queue@TiagoGeada
>>
>> CallerID: teste-geada:0:210332450:
>>
>> MaxRetries: 0
>>
>> RetryTime: 1
>>
>> WaitTime: 8640
>>
>> Account: teste-geada
>>
>> Context: TiagoGeada
>>
>> Extension: outbound
>>
>> Archive: Yes
>>
>>
>>
>>
>>
>
This will create a Local channel with two halves. The ;2 half will execute
in the dialplan at TiagoGeada,queue,1 - the ;1 half will execute in the
dialplan at TiagoGeada,outbound,1. The ;2 Local channel will execute first
until it is Answered; once Answered, that will trigger the ;1 half to start
execution. That will create two CDRs, one for each Local channel half.

MASTER_CHANNEL won't apply here, as MASTER_CHANNEL only applies to a
Parent/Child relationship between channels, that is, when one channel has
created another channel. This occurs when a channel dials another channel.
The ;1 side didn't create the ;2 side, they are effectively two sides of
the same "channel".



>
>>
>>
>>
>> ## dialplan
>>
>>
>>
>> queue => {
>>
>> Set(CDR(remoteUid)=foo);
>>
>> Queue(TiagoGeada,t,,,100);
>>
>> Hangup();
>>
>> }
>>
>>
>>
>> outbound => {
>>
>> //NoCDR();
>>
>> //ForkCDR(vdD);
>>
>> //ResetCDR(v);
>>
>> Set(CDR(remoteUid,r)=bar);
>>
>> Dial(Local/932485457@outbound,,gT);
>>
>> Hangup();
>>
>> }
>>
>>
>>
Looking at your Dialplan for the outbound extension, you dial yet another
Local channel. I would expect this to result in 3 CDR entries:

Source Channel Destination Channel
Local/queue@TiagoGeada;2
Local/queue@TiagoGeada;1   Local/932485427@outbound;1
Local/932485457@outbound;2

So, the question is, which CDR are you talking about? What value do you
want where? Keep in mind that unless all channels are answered, they won't
show up in your CDRs (unless you have unanswered=yes set in cdr.conf).

-- 
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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] WSS over Asterisk

[Matthew Jordan]

On Wed, Jun 11, 2014 at 2:58 AM, Steve Ng  wrote:

> Hi,
>
> Have anyone tried using SIPML5 to connect to Asterisk over wss?
>
> I'm having the error as shown below
>
> Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1
>  ==stack event = starting SIPml-api.js?svn=224:1
>  __tsip_transport_ws_onerror SIPml-api.js?svn=224:1
>  __tsip_transport_ws_onclose SIPml-api.js?svn=224:1
>  ==stack event = failed_to_start
>
>
> Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works
> fine. Any idea why?
>
>
There was a bug in secure WebSockets (tracked under ASTERISK-21930) that
was fixed in Asterisk 11.9.0:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.9.0-summary.html

Which version of Asterisk are you using? Is it 11.9.0 or later?

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Re: [asterisk-users] WSS over Asterisk

[William Hetherington]

Chrome 35 broke all of this you need to be using DTLS now I believe.

I had working secure web sockets with asterisk 12.2.x and chrome 34 and
then google broke eveything :)

I have not yet got around to test out DTLS etc. with chrome 35

Just so I don't waste too much time when I go to test, does anyone know if
all that's required for DTLS on the asterisk side is the following in
sip.conf?

dtlsenable=yes
dtlsverify=yes
dtlsrekey=60
dtlscafile=/usr/local/share/ca-certificates/myCA.crt
dtlscertfile=/etc/ssl/mycert.com.pem
dtlssetup=actpass

I assume I also need TLS configs in http.conf

William Hetherington
w - www.willwh.com
t - @wmwh


On Wed, Jun 11, 2014 at 11:28 AM, Matthew Jordan  wrote:

>
>
>
> On Wed, Jun 11, 2014 at 2:58 AM, Steve Ng  wrote:
>
>> Hi,
>>
>> Have anyone tried using SIPML5 to connect to Asterisk over wss?
>>
>> I'm having the error as shown below
>>
>> Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1
>>  ==stack event = starting SIPml-api.js?svn=224:1
>>  __tsip_transport_ws_onerror SIPml-api.js?svn=224:1
>>  __tsip_transport_ws_onclose SIPml-api.js?svn=224:1
>>  ==stack event = failed_to_start
>>
>>
>> Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works
>> fine. Any idea why?
>>
>>
> There was a bug in secure WebSockets (tracked under ASTERISK-21930) that
> was fixed in Asterisk 11.9.0:
>
>
> http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.9.0-summary.html
>
> Which version of Asterisk are you using? Is it 11.9.0 or later?
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Asterisk 12 AMI Hold Event

[David Huebner]

I'm trying to capture when a call is placed on and removed from being on hold 
through the AMI in Asterisk 12.3.  In previous versions, the Hold event 
contained a 'Status' field which indicated if the call was going 'On' or 'Off' 
hold.  However, in 12 not only am I not seeing the Status field, but I am not 
seeing any AMI Hold event that corresponds to removing the call from Hold.

Is this the intended behavior, or am I missing something in how the call's hold 
status should be tracked via the AMI?


Thanks!
David Huebner
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Re: [asterisk-users] WSS over Asterisk

[Matthew Jordan]

On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington 
wrote:

> Chrome 35 broke all of this you need to be using DTLS now I believe.
>
> I had working secure web sockets with asterisk 12.2.x and chrome 34
> and then google broke eveything :)
>
> I have not yet got around to test out DTLS etc. with chrome 35
>
> Just so I don't waste too much time when I go to test, does anyone know if
> all that's required for DTLS on the asterisk side is the following in
> sip.conf?
>
> dtlsenable=yes
> dtlsverify=yes
> dtlsrekey=60
> dtlscafile=/usr/local/share/ca-certificates/myCA.crt
> dtlscertfile=/etc/ssl/mycert.com.pem
> dtlssetup=actpass
>
> I assume I also need TLS configs in http.conf
>
>
Signalling is independent of the media; DTLS only affects the media.

However, there are known issues with Chrome's negotiation of DTLS and
Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961


-- 
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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] Asterisk 12 AMI Hold Event

[Richard Mudgett]

On Wed, Jun 11, 2014 at 1:35 PM, David Huebner <
david.hueb...@bolderthinking.com> wrote:

> I'm trying to capture when a call is placed on and removed from being on
> hold through the AMI in Asterisk 12.3.  In previous versions, the Hold
> event contained a 'Status' field which indicated if the call was going 'On'
> or 'Off' hold.  However, in 12 not only am I not seeing the Status field,
> but I am not seeing any AMI Hold event that corresponds to removing the
> call from Hold.
>
> Is this the intended behavior, or am I missing something in how the call's
> hold status should be tracked via the AMI?
>

This was changed in v12.  There is a Hold and an Unhold event.
Events no longer have a subevent type field.  Instead there are
separate events.  e.g., DialBegin/DialEnd, Hold/Unhold,
MusicOnHoldStart/MusicOnHoldStop, etc.

See https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+AMI+Events

Richard
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Re: [asterisk-users] Asterisk 12 AMI Hold Event

[Eric Wieling]

Generally in the UPGRADE.txt file which came in the tarball.   A pretty version 
is here https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Mudgett
Sent: Wednesday, June 11, 2014 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 12 AMI Hold Event



On Wed, Jun 11, 2014 at 1:35 PM, David Huebner 
mailto:david.hueb...@bolderthinking.com>> 
wrote:
I'm trying to capture when a call is placed on and removed from being on hold 
through the AMI in Asterisk 12.3.  In previous versions, the Hold event 
contained a 'Status' field which indicated if the call was going 'On' or 'Off' 
hold.  However, in 12 not only am I not seeing the Status field, but I am not 
seeing any AMI Hold event that corresponds to removing the call from Hold.

Is this the intended behavior, or am I missing something in how the call's hold 
status should be tracked via the AMI?

This was changed in v12.  There is a Hold and an Unhold event.
Events no longer have a subevent type field.  Instead there are
separate events.  e.g., DialBegin/DialEnd, Hold/Unhold,
MusicOnHoldStart/MusicOnHoldStop, etc.
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+AMI+Events

Richard

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Re: [asterisk-users] WSS over Asterisk

[Miguel Molina]

El 11/06/2014 1:52 p. m., Matthew Jordan escribió:




On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington > wrote:


Chrome 35 broke all of this you need to be using DTLS now I
believe.

I had working secure web sockets with asterisk 12.2.x and chrome
34 and then google broke eveything :)

I have not yet got around to test out DTLS etc. with chrome 35

Just so I don't waste too much time when I go to test, does anyone
know if all that's required for DTLS on the asterisk side is the
following in sip.conf?

dtlsenable=yes
dtlsverify=yes
dtlsrekey=60
dtlscafile=/usr/local/share/ca-certificates/myCA.crt
dtlscertfile=/etc/ssl/mycert.com.pem
dtlssetup=actpass

I assume I also need TLS configs in http.conf


Signalling is independent of the media; DTLS only affects the media.

However, there are known issues with Chrome's negotiation of DTLS and 
Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961



--
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


It is broken in Chrome (firefox never had SDES) because the WebRTC 
standard favoured the DTLS SRTP implementation instead of the SDES one. 
The thing is that although Asterisk supports DTLS implementation, it 
only supports SHA-1 hashing but both Firefox and Chrome work with 
SHA-256. The patch proposed in ASTERISK-22961 is an effort to solve this 
issue.


Best regards

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de quien las emite y no necesariamente reflejan la posicion de Millenium Phone 
Center S.A, ni comprometen la responsabilidad institucional por el uso que el 
destinatario haga de las mismas. 
- 
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Re: [asterisk-users] WSS over Asterisk

[Steve Ng]

I am using Asterisk v12.3.

As far as DTLS, I understand that applying the following Javascript will
temporarily fix for SIPML5 to Asterisk:
https://gist.github.com/steve-ng/14b9b88af43f92db1e46

WS works for me, its just wss which I'm stuck currently.


On Thu, Jun 12, 2014 at 4:37 AM, Miguel Molina <
mfmolina-lis...@millenium.com.co> wrote:

>  El 11/06/2014 1:52 p. m., Matthew Jordan escribió:
>
>
>
>
> On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington 
> wrote:
>
>> Chrome 35 broke all of this you need to be using DTLS now I believe.
>>
>>  I had working secure web sockets with asterisk 12.2.x and chrome 34
>> and then google broke eveything :)
>>
>>  I have not yet got around to test out DTLS etc. with chrome 35
>>
>>  Just so I don't waste too much time when I go to test, does anyone know
>> if all that's required for DTLS on the asterisk side is the following in
>> sip.conf?
>>
>>  dtlsenable=yes
>> dtlsverify=yes
>> dtlsrekey=60
>> dtlscafile=/usr/local/share/ca-certificates/myCA.crt
>> dtlscertfile=/etc/ssl/mycert.com.pem
>> dtlssetup=actpass
>>
>>  I assume I also need TLS configs in http.conf
>>
>>
>  Signalling is independent of the media; DTLS only affects the media.
>
> However, there are known issues with Chrome's negotiation of DTLS and
> Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961
>
>
> --
>  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
>
>
>  It is broken in Chrome (firefox never had SDES) because the WebRTC
> standard favoured the DTLS SRTP implementation instead of the SDES one. The
> thing is that although Asterisk supports DTLS implementation, it only
> supports SHA-1 hashing but both Firefox and Chrome work with SHA-256. The
> patch proposed in ASTERISK-22961 is an effort to solve this issue.
>
> Best regards
>
> --
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