[asterisk-users] Non-matching linkedid on CDR Records [SEC=UNCLASSIFIED]

[Calum Power]

Hi asterisk-users,

We have recently moved to the 13.x branch of Asterisk from 11.x, and we're 
trying to correlate CDR records from multiple-legs for billing purposes.
As part of this change we have added 'linkedid' to our CDR table schema in an 
attempt to join the multiple records into one billable record.

The call path can be simplified as (transport types in brackets):
SIP Phone---(SIP)---Asterisk Svr1---(IAX)---Asterisk Svr2---(IAX)---Asterisk 
Svr3---(SIP)---PSTN

As per the CDR spec, I expected the 'linkedid' to propagate between the records 
and be the same throughout...
However, we observe differing linkedid values for the same call... Thus (xxx's 
added for privacy):
| id  | calldate| src| dst| accountcode | 
uniqueid  | linkedid | billsec | duration |
| 5031920 | 2018-10-03 01:51:41 | 0362xx | 0449xx | | 
1538531501.18974  | 1538531501.18974 |  64 |   71 |
| 5031921 | 2018-10-03 01:52:52 | 6613   | 0449xx | 759553  | 
1538531488.11368  | 1538531488.11368 |   0 |0 |

Noting that our out-dial from Svr2 changes the CLID to 0362xx as we must 
present that CLID to our PSTN upstream provider as per their requirements.

The first record is taken from Asterisk Svr2, the second from Asterisk Svr1 
(Svr1 replicates MySQL to Svr2)
As you can see, the linkedid records are different (1538531501.18974 vs 
1538531488.11368)

The difference appears to be the matter of microsecs that it takes to connect 
the call legs (over a satellite connection) so I could probably 'guess' that 
these two are the same call, however for billing purposes this is not accurate 
enough.

Can someone shed some light on why the linkedid is not being shared between IAX 
channels?

Cheers,
Calum


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Re: [asterisk-users] WebRTC as Softphone substitute ?

[David P]

Thanks for sharing this, Alex. It sounds like TURN, as a media repeater,
wouldn't work if the media must be secured (via SRTP). Is that right?

On Wed, 3 Oct 2018, 3:17 pm alex epshteyn,  wrote:

> WebRTC requires a few specific things to be in place. We have blog posts
> that talk about WebRTC based Thirdlane Connect, but most of the information
> applies to WebRTC applications in general.
>
> https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect
>
> https://www.thirdlane.com/blog/nat-stun-turn-and-ice
>
> Best regards,
> Alex
>
>
> Alex Epshteyn
> a...@thirdlane.com
> +1 (415) 261 6601
> www.thirdlane.com
>
>
> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal  wrote:
>
> @Olivior
> I agree that seting up WebRTC is hard, however when done it is smooth to
> use. For replication you can build RPMs with working configurations.
>
> Regarding stability, it is not being used widly, so can't say it is
> mature. However we have no complain so far regarding audio or connectivity.
> sometime we provide support for "allow media / mic" type issues, but you
> know it is security feature and not a bug.
>
> Regards
>
> On Tue, Oct 2, 2018, 13:03 Olivier  wrote:
>
>> @Nasir:
>> Thanks for replying here.
>>
>> Did you met in your deployments, the kind of stability issues Carlos
>> reported earlier ?
>>
>> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  a
>> écrit :
>>
>>> Hi Olivior,
>>>
>>> We have recently worked on a WebRTC based agent panel. As based on my
>>> experience I think that WebRTC based phones are far better and cheaper then
>>> those soft / sip phone. the big plus is that they are easy to customize and
>>> developer can use the power of browser and web to build / offer features
>>> which are not possible with regular phones.
>>>
>>> Regarding your concern about BLF or call history, for me as a being
>>> developer it is just a matter of customization.
>>>
>>> Regards
>>>
>>> Nasir Iqbal
>>>
>>> ICTBroadcast - an Auto Dialer software for ITSP
>>> 
>>> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
>>> http://www.ictbroadcast.com/
>>>
>>>
>>> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez 
>>> wrote:
>>>
 On 9/26/18 10:20 AM, Matthew Fredrickson wrote:

 > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez 
 wrote:
 >> On 9/26/2018 4:46 AM, Olivier wrote:
 >>
 >>> Hello,
 >>>
 >>> This morning, I asked myself if WebRTC could be a viable alternative
 >>> to softphone deployment.
 >>>
 >>> For me, main issue with Softphones is the amount of work needed for
 >>> installation and configuration.
 >>> Also, Softphones must be carefully choosen if Deskphone-like quality
 >>> is expected.
 >>>
 >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
 >>> Softphone features (call history, BLF, ...) for WebRTC deployment
 >>> simplicity.
 >>>
 >>> What do you think of this ?
 >>> What kind of experience did you met with such WebRTC deployments ?
 >>> What about classic telephony features (CallTransfer) ?
 >>> Have you tried Cyber Maga Phone 2K ?
 >>>
 >>   If you can get it to work WebRTC is a good option.  The
 problem is
 >> that any changes in your network may disrupt it and even trying to
 >> replicate your installation is difficult.  I have it working fine on
 my
 >> website so customers can call us directly from our web page but I
 never
 >> could get Cyber Mega Phone 2K to work on the same server.  We used
 JSSIP
 >> to create the webrtc phone on our website.
 > We just updated the documentation for how to get CMP2K working on the
 > wiki [1].  We'd love some feedback if you still have issues getting it
 > setup so that we can improve the docs.
 >
 > [1]
 https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
 >
 > Best wishes,
 > Matthew Fredrickson
 >
  I followed the procedure indicated in the link but I cannot get
 remote video.  I can only see my own feed.  We do have audio for a
 little while.  For some reason the users get disconnected after a few
 minutes even though you can still see your video feed on screen.  This
 was done with Asterisk 15.6.0

 --
 Telecomunicaciones Abiertas de México S.A. de C.V.
 Carlos Chávez
 +52 (55)8116-9161


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

 Astricon is coming up October 9-11!  Signup is available at:
 https://www.asterisk.org/community/astricon-user-conference

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

 New to Asterisk? Start here:
   https://wiki.asterisk.org/wiki/display

Re: [asterisk-users] WebRTC as Softphone substitute ?

[alex epshteyn]

WebRTC requires a few specific things to be in place. We have blog posts that 
talk about WebRTC based Thirdlane Connect, but most of the information applies 
to WebRTC applications in general.

https://www.thirdlane.com/blog/get-up-and-running-with-thirdlane-connect

https://www.thirdlane.com/blog/nat-stun-turn-and-ice

Best regards,
Alex


Alex Epshteyn
a...@thirdlane.com
+1 (415) 261 6601
www.thirdlane.com



> On Oct 2, 2018, at 6:08 PM, Nasir Iqbal  wrote:
> 
> @Olivior
> I agree that seting up WebRTC is hard, however when done it is smooth to use. 
> For replication you can build RPMs with working configurations.
> 
> Regarding stability, it is not being used widly, so can't say it is mature. 
> However we have no complain so far regarding audio or connectivity. sometime 
> we provide support for "allow media / mic" type issues, but you know it is 
> security feature and not a bug.
> 
> Regards
> 
> On Tue, Oct 2, 2018, 13:03 Olivier  > wrote:
> @Nasir:
> Thanks for replying here.
> 
> Did you met in your deployments, the kind of stability issues Carlos reported 
> earlier ?
> 
> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  > a écrit :
> Hi Olivior,
> 
> We have recently worked on a WebRTC based agent panel. As based on my 
> experience I think that WebRTC based phones are far better and cheaper then 
> those soft / sip phone. the big plus is that they are easy to customize and 
> developer can use the power of browser and web to build / offer features 
> which are not possible with regular phones. 
> 
> Regarding your concern about BLF or call history, for me as a being developer 
> it is just a matter of customization.
> 
> Regards
> 
> Nasir Iqbal
> 
> ICTBroadcast - an Auto Dialer software for ITSP 
> 
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/ 
> 
> 
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez  > wrote:
> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
> 
> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez  > > wrote:
> >> On 9/26/2018 4:46 AM, Olivier wrote:
> >>
> >>> Hello,
> >>>
> >>> This morning, I asked myself if WebRTC could be a viable alternative
> >>> to softphone deployment.
> >>>
> >>> For me, main issue with Softphones is the amount of work needed for
> >>> installation and configuration.
> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
> >>> is expected.
> >>>
> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
> >>> simplicity.
> >>>
> >>> What do you think of this ?
> >>> What kind of experience did you met with such WebRTC deployments ?
> >>> What about classic telephony features (CallTransfer) ?
> >>> Have you tried Cyber Maga Phone 2K ?
> >>>
> >>   If you can get it to work WebRTC is a good option.  The problem is
> >> that any changes in your network may disrupt it and even trying to
> >> replicate your installation is difficult.  I have it working fine on my
> >> website so customers can call us directly from our web page but I never
> >> could get Cyber Mega Phone 2K to work on the same server.  We used JSSIP
> >> to create the webrtc phone on our website.
> > We just updated the documentation for how to get CMP2K working on the
> > wiki [1].  We'd love some feedback if you still have issues getting it
> > setup so that we can improve the docs.
> >
> > [1] 
> > https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
> >  
> > 
> >
> > Best wishes,
> > Matthew Fredrickson
> >
>  I followed the procedure indicated in the link but I cannot get 
> remote video.  I can only see my own feed.  We do have audio for a 
> little while.  For some reason the users get disconnected after a few 
> minutes even though you can still see your video feed on screen.  This 
> was done with Asterisk 15.6.0
> 
> -- 
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez
> +52 (55)8116-9161
> 
> 
> -- 
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com 
>  --
> 
> Astricon is coming up October 9-11!  Signup is available at: 
> https://www.asterisk.org/community/astricon-user-conference 
> 
> 
> Check out the new Asterisk community forum at: 
> https://community.asterisk.org/ 
> 
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started 
> 

Re: [asterisk-users] WebRTC as Softphone substitute ?

[Nasir Iqbal]

@Olivior
I agree that seting up WebRTC is hard, however when done it is smooth to
use. For replication you can build RPMs with working configurations.

Regarding stability, it is not being used widly, so can't say it is mature.
However we have no complain so far regarding audio or connectivity.
sometime we provide support for "allow media / mic" type issues, but you
know it is security feature and not a bug.

Regards

On Tue, Oct 2, 2018, 13:03 Olivier  wrote:

> @Nasir:
> Thanks for replying here.
>
> Did you met in your deployments, the kind of stability issues Carlos
> reported earlier ?
>
> Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  a
> écrit :
>
>> Hi Olivior,
>>
>> We have recently worked on a WebRTC based agent panel. As based on my
>> experience I think that WebRTC based phones are far better and cheaper then
>> those soft / sip phone. the big plus is that they are easy to customize and
>> developer can use the power of browser and web to build / offer features
>> which are not possible with regular phones.
>>
>> Regarding your concern about BLF or call history, for me as a being
>> developer it is just a matter of customization.
>>
>> Regards
>>
>> Nasir Iqbal
>>
>> ICTBroadcast - an Auto Dialer software for ITSP
>> 
>> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
>> http://www.ictbroadcast.com/
>>
>>
>> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez 
>> wrote:
>>
>>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
>>>
>>> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez 
>>> wrote:
>>> >> On 9/26/2018 4:46 AM, Olivier wrote:
>>> >>
>>> >>> Hello,
>>> >>>
>>> >>> This morning, I asked myself if WebRTC could be a viable alternative
>>> >>> to softphone deployment.
>>> >>>
>>> >>> For me, main issue with Softphones is the amount of work needed for
>>> >>> installation and configuration.
>>> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
>>> >>> is expected.
>>> >>>
>>> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
>>> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
>>> >>> simplicity.
>>> >>>
>>> >>> What do you think of this ?
>>> >>> What kind of experience did you met with such WebRTC deployments ?
>>> >>> What about classic telephony features (CallTransfer) ?
>>> >>> Have you tried Cyber Maga Phone 2K ?
>>> >>>
>>> >>   If you can get it to work WebRTC is a good option.  The problem
>>> is
>>> >> that any changes in your network may disrupt it and even trying to
>>> >> replicate your installation is difficult.  I have it working fine on
>>> my
>>> >> website so customers can call us directly from our web page but I
>>> never
>>> >> could get Cyber Mega Phone 2K to work on the same server.  We used
>>> JSSIP
>>> >> to create the webrtc phone on our website.
>>> > We just updated the documentation for how to get CMP2K working on the
>>> > wiki [1].  We'd love some feedback if you still have issues getting it
>>> > setup so that we can improve the docs.
>>> >
>>> > [1]
>>> https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
>>> >
>>> > Best wishes,
>>> > Matthew Fredrickson
>>> >
>>>  I followed the procedure indicated in the link but I cannot get
>>> remote video.  I can only see my own feed.  We do have audio for a
>>> little while.  For some reason the users get disconnected after a few
>>> minutes even though you can still see your video feed on screen.  This
>>> was done with Asterisk 15.6.0
>>>
>>> --
>>> Telecomunicaciones Abiertas de México S.A. de C.V.
>>> Carlos Chávez
>>> +52 (55)8116-9161
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Astricon is coming up October 9-11!  Signup is available at:
>>> https://www.asterisk.org/community/astricon-user-conference
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Astricon is coming up October 9-11!  Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> __

[asterisk-users] Call Queue Data

[Tech Support]

All;

A few years back, we put a heck of a lot of effort into developing a
software package to analyze call queue data that we want to open source.
It's a pretty good package and I would like to dust it off and resurrect it.
What I need to do that is have sample call queue data to test with. If
anyone has queue data they would be able to send me, I would be very, very
grateful to them. And I give people my word that if they can send some data,
I would immediately anonymize it. If that's not possible, then how can I
generate sample queue test data to work with? Are there any packages out
there that can generate this?

Thanks in Advance;

John V.

 

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Re: [asterisk-users] WebRTC as Softphone substitute ?

[Olivier]

@Nasir:
Thanks for replying here.

Did you met in your deployments, the kind of stability issues Carlos
reported earlier ?

Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal  a
écrit :

> Hi Olivior,
>
> We have recently worked on a WebRTC based agent panel. As based on my
> experience I think that WebRTC based phones are far better and cheaper then
> those soft / sip phone. the big plus is that they are easy to customize and
> developer can use the power of browser and web to build / offer features
> which are not possible with regular phones.
>
> Regarding your concern about BLF or call history, for me as a being
> developer it is just a matter of customization.
>
> Regards
>
> Nasir Iqbal
>
> ICTBroadcast - an Auto Dialer software for ITSP
> 
> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns
> http://www.ictbroadcast.com/
>
>
> On Thu, Sep 27, 2018 at 1:06 AM Carlos Chavez  wrote:
>
>> On 9/26/18 10:20 AM, Matthew Fredrickson wrote:
>>
>> > On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez 
>> wrote:
>> >> On 9/26/2018 4:46 AM, Olivier wrote:
>> >>
>> >>> Hello,
>> >>>
>> >>> This morning, I asked myself if WebRTC could be a viable alternative
>> >>> to softphone deployment.
>> >>>
>> >>> For me, main issue with Softphones is the amount of work needed for
>> >>> installation and configuration.
>> >>> Also, Softphones must be carefully choosen if Deskphone-like quality
>> >>> is expected.
>> >>>
>> >>> Now that WebRTC becomes ubiquitous, it might make sense to trade
>> >>> Softphone features (call history, BLF, ...) for WebRTC deployment
>> >>> simplicity.
>> >>>
>> >>> What do you think of this ?
>> >>> What kind of experience did you met with such WebRTC deployments ?
>> >>> What about classic telephony features (CallTransfer) ?
>> >>> Have you tried Cyber Maga Phone 2K ?
>> >>>
>> >>   If you can get it to work WebRTC is a good option.  The problem
>> is
>> >> that any changes in your network may disrupt it and even trying to
>> >> replicate your installation is difficult.  I have it working fine on my
>> >> website so customers can call us directly from our web page but I never
>> >> could get Cyber Mega Phone 2K to work on the same server.  We used
>> JSSIP
>> >> to create the webrtc phone on our website.
>> > We just updated the documentation for how to get CMP2K working on the
>> > wiki [1].  We'd love some feedback if you still have issues getting it
>> > setup so that we can improve the docs.
>> >
>> > [1]
>> https://wiki.asterisk.org/wiki/display/AST/Installing+and+Configuring+CyberMegaPhone
>> >
>> > Best wishes,
>> > Matthew Fredrickson
>> >
>>  I followed the procedure indicated in the link but I cannot get
>> remote video.  I can only see my own feed.  We do have audio for a
>> little while.  For some reason the users get disconnected after a few
>> minutes even though you can still see your video feed on screen.  This
>> was done with Asterisk 15.6.0
>>
>> --
>> Telecomunicaciones Abiertas de México S.A. de C.V.
>> Carlos Chávez
>> +52 (55)8116-9161
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Astricon is coming up October 9-11!  Signup is available at:
>> https://www.asterisk.org/community/astricon-user-conference
>>
>> Check out the new Asterisk community forum at:
>> https://community.asterisk.org/
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
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Astricon is coming up October 9-11!  Signup is available at: 
https://www.asterisk.org/community/astricon-user-conference

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

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

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To UNSUBSCRIBE or update options visit:
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