[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=217626#comment-217626 ] 

armeniki edited comment on PRI-168 at 4/25/14 10:14 PM:
--------------------------------------------------------

Hi Richard,

Thanks for your reply.  Sorry, it's been a while and I didn't know we can't submit new feature requests any longer... although the system still has the "New Feature" option when submitting the request type?

Anyways, I've tried what you have suggested already and still no luck.  It's really frustrating because we can only get two outcomes:

1) A user dials a busy number and they see the "User busy" message on their phone for a few seconds, they never hear a busy tone or any audio, and the call disconnects.

or

2) A user dials a busy number and they can hear the busy signal or audio, but they never see a message "user busy".

I realise this may seem trivial but our congestion/reorder signal and busy signals here are very similar and lots of people can't tell the difference so that's why we need to have the cause code displayed on the user's phone and congestion or busy audio played at the same time.

On our production system which is a Nortel BCM 400 connected to our telco's E1, this happens flawlessly.. a user will dial a number and they will hear the audio tone/message and the phone's display wills how the cause code for about 5-8 seconds.  All we are trying to do is get Asterisk to work in the same manner.  

It seems that whatever I do with chan_dahdi or extensions.conf I can only get one or the other.... and that Asterisk always "spawns the extension" when sending out the PRI code.  

Originally, I had thought that I could send the code via Progress after setting the PRI_CAUSE variable but that's not how it works I presume.

I'm wondering if the code for the Busy() or Congestion() functions can be modified so they don't spawn the channel when they finish up and the next entry in the dialplan/context is executed instead giving a change for a recording to be played to the caller.. this would be interesting.

I do have programming experience but it's been a while... and I'm just dreading having to go through the code to see if I can make changes... so yup, that's where we are at at the moment.

Would appreciate any further insight.

Cheers,

Armen
Sydney, New South Wales, Australia



was (Author: armeniki):
Hi Richard,

Thanks for your reply.  Sorry, it's been a while and I didn't know we can't submit new feature requests any longer... although the system still has the "New Feature" option when submitting the request type?

Anyways, I've tried what you have suggested already and still no luck.  It's really frustrating because we can only get two outcomes:

1) A user dials a busy number and they see the "User busy" message on their phone for a few seconds, they never hear a busy tone or any audio, and the call disconnects.

or

2) A user dials a busy number and they can hear the busy signal or audio, but they never see a message "user busy".

I realise this may seem trivial but our congestion/reorder signal and busy signals here are very similar and lots of people can't tell the difference so that's why we need to have the cause code displayed on the user's phone and congestion or busy audio played at the same time.

On our production system which is a Nortel BCM 400 connected to our telco's E1, this happens flawlessly.. a user will dial a number and they will hear the audio tone/message and the phone's display wills how the cause code for about 5-8 seconds.  All we are trying to do is get Asterisk to work in the same manner.  

It seems that whatever I do with chan_dahdi or extensions.conf I can only get one or the other.... and that Asterisk always "spawns the extension" when sending out the PRI code.  

Originally, I had thought that I could send the code via Progress after setting the PRI_CAUSE variable but that's not how it works I presume.

I do have programming experience but it's been a while... and I'm just dreading having to go through the code to see if I can make changes... so yup, that's where we are at at the moment.

Would appreciate any further insight.

Cheers,

Armen
Sydney, New South Wales, Australia


> 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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