Re: [asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Ø  Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
>Does that have to be set in the Dial handler in order to get set on the 
>dialled channel or can it be inheritted?

Never mind. Tested it. Working great! Thanks.
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Re: [asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Ø  Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
Does that have to be set in the Dial handler in order to get set on the dialled 
channel or can it be inheritted?

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Re: [asterisk-users] Beep on Attended Transfer

[Richard Mudgett]

On Thu, Feb 16, 2017 at 2:02 PM, Daniel Journo  wrote:

> Hi,
>
>
>
> During an attended transfer using the SIP phone feature buttons, I’m
> getting a few complaints from recipients that they can’t tell when the call
> they are receiving has been transferred.
>
> Is there any way (even if it’s complicated) to generate a beep tone to the
> recipient of the transferred call when the transfer is completed?
>
>
>
> I know you can do this with DTMF codes but they want to use the phone’s
> transfer features.
>

Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.

Richard
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[asterisk-users] Beep on Attended Transfer

[Daniel Journo]

Hi,

During an attended transfer using the SIP phone feature buttons, I'm getting a 
few complaints from recipients that they can't tell when the call they are 
receiving has been transferred.
Is there any way (even if it's complicated) to generate a beep tone to the 
recipient of the transferred call when the transfer is completed?

I know you can do this with DTMF codes but they want to use the phone's 
transfer features.

Many thanks
Dan

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Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

[Matthew Jordan]

On Thu, Feb 16, 2017 at 9:05 AM, Olivier  wrote:

>
>
> 2017-02-16 14:27 GMT+01:00 Joshua Colp :
>
>> On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote:
>> > Hello,
>> >
>> > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP
>> > hardphone.
>> >
>> > When a phone has enabled this feature, it would send a SIP PUBLISH to
>> its
>> > SIP Server letting this server dispatch to whatever is needs to.
>> >
>> > These messages are sent during calls but may also be sent when a call is
>> > over.
>> >
>> > At the moment, I'm using Asterisk to serve these SIP phones so my
>> > Asterisk
>> > box receives those SIP PUBLISH and discard them with a 489 Bad Event
>> > reply.
>> >
>> > I'm not using or planning to use any Kamalio server.
>> >
>> > 1. Is there an Asterisk version that would allow me to read (and store)
>> > in
>> > or out-of-band SIP PUBLISH messages from SIP phones ?
>> > 2. Alternatively, is there an Asterisk version that would allow me to
>> > relay
>> > those messages somewhere ?
>>
>> No version of Asterisk allows the handling or relaying of these
>> arbitrary PUBLISH messages. In the case of PJSIP though that is
>> pluggable so a C module could be written to do something.
>>
>
> From RFC 6035, "This document defines a new SIP event package, vq-rtcpxr,
> and
> a new MIME type, application/vq-rtcpxr, that enable the collection and
> reporting of metrics".
>
> As I'm not aware of many SIP event package currently implemented in
> PJSIP/Asterisk acting for
> out-of-calls events, it shouldn't be easy to mimic current features to add
> this new one.
>
>
>
>> > 3. Would a Kamalio-like box allow me to do this ?
>>
>> You could act as you wish on the PUBLISH requests in Kamailio.
>>
>
> This seams easier, for the moment.
>
> I think I still need to better understand what are mixed Asterisk-Kamailio
> architectures main strengths
> compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that
> is another story.
>
> Thank you very much for replying.
>
>
This admittedly high speed presentation that glosses over lots of complex
topics may or may not help you:

http://ftp.osuosl.org/pub/fosdem/2017/K.3.401/asterisk.mp4

/shameless plug off

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Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

[Olivier]

2017-02-16 14:27 GMT+01:00 Joshua Colp :

> On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote:
> > Hello,
> >
> > I'm currently testing a so-called VQ RTCP-XR feature from a a SIP
> > hardphone.
> >
> > When a phone has enabled this feature, it would send a SIP PUBLISH to its
> > SIP Server letting this server dispatch to whatever is needs to.
> >
> > These messages are sent during calls but may also be sent when a call is
> > over.
> >
> > At the moment, I'm using Asterisk to serve these SIP phones so my
> > Asterisk
> > box receives those SIP PUBLISH and discard them with a 489 Bad Event
> > reply.
> >
> > I'm not using or planning to use any Kamalio server.
> >
> > 1. Is there an Asterisk version that would allow me to read (and store)
> > in
> > or out-of-band SIP PUBLISH messages from SIP phones ?
> > 2. Alternatively, is there an Asterisk version that would allow me to
> > relay
> > those messages somewhere ?
>
> No version of Asterisk allows the handling or relaying of these
> arbitrary PUBLISH messages. In the case of PJSIP though that is
> pluggable so a C module could be written to do something.
>

>From RFC 6035, "This document defines a new SIP event package, vq-rtcpxr,
and
a new MIME type, application/vq-rtcpxr, that enable the collection and
reporting of metrics".

As I'm not aware of many SIP event package currently implemented in
PJSIP/Asterisk acting for
out-of-calls events, it shouldn't be easy to mimic current features to add
this new one.



> > 3. Would a Kamalio-like box allow me to do this ?
>
> You could act as you wish on the PUBLISH requests in Kamailio.
>

This seams easier, for the moment.

I think I still need to better understand what are mixed Asterisk-Kamailio
architectures main strengths
compared to alternatives (Asterisk alone, Kamailio/RTPproxy, ...) but that
is another story.

Thank you very much for replying.



>
> --
> 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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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Am 16.02.2017 um 15:01 schrieb Joshua Colp:

> As for your issues please do file them. I'd also suggest using bundled
> PJSIP, it works the best with Asterisk and we backport applicable fixes
> and include fixes we've created that have not yet made it to a PJSIP
> release.

OK, I'll try again with the bundled version.
If the bugs persist, I'll file some bugs ;-)


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Joshua Colp]

On Thu, Feb 16, 2017, at 09:51 AM, Max Grobecker wrote:
> Hi,
> 
> Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> > And Microsip using PJSIP SIP stack :)
> 
> Sorry (also, for off-topic), based on my latest experience with PJSIP,
> I'm not sure if this really is a sign of good quality.
> Maybe it's a problem with the implementation in Asterisk (I haven't tried
> PJSIP in other software), but after just five minutes of testing
> I found several bugs regarding PJSIP preventing me to use it in a
> production enviroment :-(
> I'm going to file these bugs at the moment...

Asterisk uses PJSIP at a different layer than a lot of other things. A
lot of the application logic and features are done by Asterisk, while
clients based on PJSIP use pjsua which takes care of that. You can't
compare the two.

As for your issues please do file them. I'd also suggest using bundled
PJSIP, it works the best with Asterisk and we backport applicable fixes
and include fixes we've created that have not yet made it to a PJSIP
release.

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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hi,

Am 16.02.2017 um 14:19 schrieb Annus Fictus:
> And Microsip using PJSIP SIP stack :)

Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not 
sure if this really is a sign of good quality.
Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP 
in other software), but after just five minutes of testing
I found several bugs regarding PJSIP preventing me to use it in a production 
enviroment :-(
I'm going to file these bugs at the moment...


Max



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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[A J Stiles]

On Thursday 16 Feb 2017, Max Grobecker wrote:
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client supports TLS, SRTP and ZRTP:
> http://phonerlite.de/features_en.htm
> 
> Yes, the GUI is not that much user friendly as Zoiper is - but at least a
> very good and stable client for testing purposes ;-)

It seems to be Windows-only, though, and does not appear to include any Source 
Code  (even although the licence allows you to distribute modified versions).  
Either of those could potentially be a show-stopper.

Linphone  ( http://www.linphone.org/technical-corner/linphone/overview )  
supports TLS -- and is both cross-platform and GPLv2.

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Re: [asterisk-users] How to read or relay SIP PUBLISH messages ?

[Joshua Colp]

On Thu, Feb 16, 2017, at 09:11 AM, Olivier wrote:
> Hello,
> 
> I'm currently testing a so-called VQ RTCP-XR feature from a a SIP
> hardphone.
> 
> When a phone has enabled this feature, it would send a SIP PUBLISH to its
> SIP Server letting this server dispatch to whatever is needs to.
> 
> These messages are sent during calls but may also be sent when a call is
> over.
> 
> At the moment, I'm using Asterisk to serve these SIP phones so my
> Asterisk
> box receives those SIP PUBLISH and discard them with a 489 Bad Event
> reply.
> 
> I'm not using or planning to use any Kamalio server.
> 
> 1. Is there an Asterisk version that would allow me to read (and store)
> in
> or out-of-band SIP PUBLISH messages from SIP phones ?
> 2. Alternatively, is there an Asterisk version that would allow me to
> relay
> those messages somewhere ?

No version of Asterisk allows the handling or relaying of these
arbitrary PUBLISH messages. In the case of PJSIP though that is
pluggable so a C module could be written to do something.

> 3. Would a Kamalio-like box allow me to do this ?

You could act as you wish on the PUBLISH requests in Kamailio.

-- 
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Annus Fictus]

And Microsip using PJSIP SIP stack :)


El 16/02/2017 a las 08:15, Jonathan H escribió:

Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/

On 16 February 2017 at 13:04, Max Grobecker
 wrote:

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:

Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
security reasons I have enable TLS on our Asterisk server to support TLS 
authentication, It works well with hard phones. Has anybody in this forum use 
SIP Soft phones with TLS authentication enabled? Any suggestions?



Thanks,
Motty





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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Jonathan H]

Microsip (Windows) is free and small.
2.5Mb download, 10Mb RAM usage, does everything I need and configuring
TLS is a doddle.
http://www.microsip.org/

On 16 February 2017 at 13:04, Max Grobecker
 wrote:
> Hello,
>
> I'm a big fan of PhonerLite.
> It's more poplar in Germany, but also available in English language.
> This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm
>
> Yes, the GUI is not that much user friendly as Zoiper is - but at least a 
> very good and stable client for testing purposes ;-)
>
> Max
>
>
> Am 15.02.2017 um 19:46 schrieb Motty Cruz:
>> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
>> security reasons I have enable TLS on our Asterisk server to support TLS 
>> authentication, It works well with hard phones. Has anybody in this forum 
>> use SIP Soft phones with TLS authentication enabled? Any suggestions?
>>
>>
>>
>> Thanks,
>> Motty
>>
>>
>>
>
>
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[asterisk-users] How to read or relay SIP PUBLISH messages ?

[Olivier]

Hello,

I'm currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone.

When a phone has enabled this feature, it would send a SIP PUBLISH to its
SIP Server letting this server dispatch to whatever is needs to.

These messages are sent during calls but may also be sent when a call is
over.

At the moment, I'm using Asterisk to serve these SIP phones so my Asterisk
box receives those SIP PUBLISH and discard them with a 489 Bad Event reply.

I'm not using or planning to use any Kamalio server.

1. Is there an Asterisk version that would allow me to read (and store) in
or out-of-band SIP PUBLISH messages from SIP phones ?
2. Alternatively, is there an Asterisk version that would allow me to relay
those messages somewhere ?
3. Would a Kamalio-like box allow me to do this ?
4. Suggestions ?

Best
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Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

[Max Grobecker]

Hello,

I'm a big fan of PhonerLite.
It's more poplar in Germany, but also available in English language.
This client supports TLS, SRTP and ZRTP: http://phonerlite.de/features_en.htm

Yes, the GUI is not that much user friendly as Zoiper is - but at least a very 
good and stable client for testing purposes ;-)

Max


Am 15.02.2017 um 19:46 schrieb Motty Cruz:
> Hello, I have a user that prefers Soft SIP phone install on his laptop, for 
> security reasons I have enable TLS on our Asterisk server to support TLS 
> authentication, It works well with hard phones. Has anybody in this forum use 
> SIP Soft phones with TLS authentication enabled? Any suggestions?
> 
>  
> 
> Thanks,
> Motty
> 
> 
> 



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Re: [asterisk-users] [OT] Downloading Recqual [SOLVED]

[Olivier]

2017-02-16 12:21 GMT+01:00 A J Stiles :

> On Thursday 16 Feb 2017, Olivier wrote:
> > Hello,
> >
> > While googling, I've just discovered Recqual.
> > If I'm not mistaken, project's sourceforge site [2] does not host any
> > source or binary.
>
> You need to follow the "code" link, copy the line that starts with
>  "svn checkout ..."
> and then just paste that straight into a terminal.  (You will need to have
> installed the "subversion" package before you start.)
>
> --
> AJS
>
> Note:  Originating address only accepts e-mail from list!  If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
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> To UNSUBSCRIBE or update options visit:
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>

Yes: I missed this Code tab completely.
Using it, I could check out the code
Using Download snapshot at the top of this Code tab also let you download a
zip file (and avoid installing svn just for that).

Thank you very much for this.
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Re: [asterisk-users] [OT] Downloading Recqual

[A J Stiles]

On Thursday 16 Feb 2017, Olivier wrote:
> Hello,
> 
> While googling, I've just discovered Recqual.
> If I'm not mistaken, project's sourceforge site [2] does not host any
> source or binary.

You need to follow the "code" link, copy the line that starts with
 "svn checkout ..."
and then just paste that straight into a terminal.  (You will need to have 
installed the "subversion" package before you start.)

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[asterisk-users] [OT] Downloading Recqual

[Olivier]

Hello,

While googling, I've just discovered Recqual.
If I'm not mistaken, project's sourceforge site [2] does not host any
source or binary.

Is there an alternative location to download this ?
Suggestions ?

Best regards


[1] http://blog.krisk.org/2008/12/introducing-recqual.html
[2] https://sourceforge.net/projects/recqual/
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