[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 edited comment on PRI-168 at 4/27/14 12:31 AM:
--------------------------------------------------------

Hi Richard,

Yes, you're right, in this case we would be *only* interested in Point 1, ie: the "sending" aspect of the DISCONNECT process.  This is because Asterisk, in this scenario is acting as the "Telco" for the BCM400.  

If there is indeed a deficiency as you mentioned, it might be worthwhile taking a look into it sometime. I looked at the ETS Euro ISDN standards document (http://www.etsi.org/deliver/etsi_i_ets/300100_300199/30010201/01_60/ets_30010201e01p.pdf) and it does mention this.  In fact, this is what the network-side timer T306 is for which is already part of the LibPRI's build.  (Page 152, Table 9.1 of the standard.)

Just to be double sure....  Nortel has a really handy tool for monitoring their PBX and the PRI connections to it, called BCM Monitor (I'm sure you know of it).  So, I had a look on both the production system connected to the Telco and the test system connected to Asterisk whilst dialling a busy number.  Both times I did not hangup, I simply waited until the network disconnected me.  

On the Telco setup, as soon as I dialed the number, the message User busy 17 came up and I heard the engaged signal.  This continued for 30 seconds.

On the Asterisk setup, as soon as I dialed the number, the message User busy 17 briefly appeared and I got disconnected straight away without hearing anything.

Here's a screen shot of the Telco connection:
http://i1100.photobucket.com/albums/g408/armeniki/telephone/ISDN-telco.png

Here's a screen shot of the Asterisk connection:
http://i1100.photobucket.com/albums/g408/armeniki/telephone/ISDN_asterisk.png

If you look towards the end, you'll notice with the Telco connection that at:

*22:20:45 > CC > Cref Origin: Cref 3  DISCONNECT User busy*
--There is a Progress Indicator attached to it..... 

However, on the Asterisk system at:
*22:27:32 > CC > Cref Origin: Cref 1  DISCONNECT User busy*
--There is no Progress Indicator attached and the PBX simply responds with the RELEASE_CC.

And this is reason why we are having this issue.. the lack of those important *03 00 81 88* hex characters coming through...


was (Author: armeniki):
Hi Richard,

Yes, you're right, in this case we would be *only* interested in Point 1, ie: the "sending" aspect of the DISCONNECT process.  This is because Asterisk, in this scenario is acting as the "Telco" for the BCM400.  

If there is indeed a deficiency as you mentioned, it might be worthwhile taking a look into it sometime. I looked at the ETS Euro ISDN standards document (http://www.etsi.org/deliver/etsi_i_ets/300100_300199/30010201/01_60/ets_30010201e01p.pdf) and it does mention this.  In fact, this is what the network-side timer T306 is for which is already part of the LibPRI's EuroISDN configuration.  (Page 152, Table 9.1 of the standard.)

Just to be double sure....  Nortel has a really handy tool for monitoring their PBX and the PRI connections to it, called BCM Monitor (I'm sure you know of it).  So, I had a look on both the production system connected to the Telco and the test system connected to Asterisk whilst dialling a busy number.  Both times I did not hangup, I simply waited until the network disconnected me.  

On the Telco setup, as soon as I dialed the number, the message User busy 17 came up and I heard the engaged signal.  This continued for 30 seconds.

On the Asterisk setup, as soon as I dialed the number, the message User busy 17 briefly appeared and I got disconnected straight away without hearing anything.

Here's a screen shot of the Telco connection:
http://i1100.photobucket.com/albums/g408/armeniki/telephone/ISDN-telco.png

Here's a screen shot of the Asterisk connection:
http://i1100.photobucket.com/albums/g408/armeniki/telephone/ISDN_asterisk.png

If you look towards the end, you'll notice with the Telco connection that at:

*22:20:45 > CC > Cref Origin: Cref 3  DISCONNECT User busy*
--There is a Progress Indicator attached to it..... 

However, on the Asterisk system at:
*22:27:32 > CC > Cref Origin: Cref 1  DISCONNECT User busy*
--There is no Progress Indicator attached and the PBX simply responds with the RELEASE_CC.

And this is reason why we are having this issue.. the lack of those important *03 00 81 88* hex characters coming through...

> 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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