[asterisk-bugs] [JIRA] (PRI-168) Alternative Mode of Sending PRI Cause Codes

[armeniki (JIRA)]

    [ https://issues.asterisk.org/jira/browse/PRI-168?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217629#comment-217629 ] 

armeniki commented on PRI-168:
------------------------------

Hi Richard,

Yes, you're right, in this case we would be interested in the "sending" aspect as the Asterisk system in this scenario is the "Telco" for the BCM400 PBX.  

If there is indeed a deficiency as you mentioned, it might be worthwhile taking a looking into getting this sorted sometime. 

In fact, I have checked the ETS Euro ISDN standards document and it does indeed have specifications for what I am referring to, not surprisingly as the Telco does this.

Here's the link to the document in case you wanted to have a look:
http://www.etsi.org/deliver/etsi_i_ets/300100_300199/30010201/01_60/ets_30010201e01p.pdf

In section 5.3.4 it describes Clearing with Tones and Announcements provided and refers to when the DISCONNECT message contains
progress indicator #8 - just as you mention in your comment above in point 1.


> Alternative Mode of Sending PRI Cause Codes
> -------------------------------------------
>
>                 Key: PRI-168
>                 URL: https://issues.asterisk.org/jira/browse/PRI-168
>             Project: LibPRI
>          Issue Type: New Feature
>      Security Level: None
>    Affects Versions: 1.4.13
>            Reporter: armeniki
>            Assignee: Richard Mudgett
>            Severity: Minor
>
> Hi everyone,
> As you know, currently when the Hangup() command is used in the Asterisk dial plan, it will tear down both the far and and near end of the call and audio amongst other things.  In addition, there is a way of indicating the PRI Cause Code to the user by entering the code within the parenthesis ie: Hangup(1) or Hangup(16), etc.
> Here is the issue:  We have a PBX connected to an E1/PRI from the Telco.  This is a production system and does not use Asterisk.  On this system, whenever someone dials a number which is busy, for example, the phone's display will show the standard PRI message "user busy" and the user will hear an engaged signal (busy signal).  Likewise, if a wrong number is dialled, the display will show "unallocated num" and another tone or message will be heard.... and so on.
> Currently, we're doing tests on the same type of system connected to Asterisk via an E1/PRI and have found that this does not happen.  Basically, if the user dials a busy number, they will hear a busy signal but they won't see the "user busy" message.  Alternatively, if we change the extension script a bit, we can issue a Hangup(17) to make the phone show "user busy" but then there's no way to play the busy signal because the phone/channel hangs up.
> SO....
> Would it be possible to perhaps create a new function called "SendPRICause()" (kind of like the SendText function for SIP phones) so that we can use that instead of Hangup()?
> This way the use can see the message and hear whatever recording needs to be played to them at the same time.
> An example of its usage could be:
> exten => s-DN_CHANGED,1,Progress()
> exten => s-DN_CHANGED,1,SendPRICause(22)
> exten => s-DN_CHANGED,n,Playback(/var/lib/asterisk/sounds/tel/sorry-number-changed,noanswer)
> exten => s-DN_CHANGED,n,Hangup()
> --------------------------
> Cheers,
> Armen



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