Asterisk PRI Logger with Standard PBX not working

[urmi.l]

I want to set up the PRI logger using Asterisk.
We are having standard PBX (Panasonic TDA00) for the PSTN phones. All the Incoming and Out going calls needs to be recorded.

Bellow is my setup :

I m having 1 Asterisk server with ATCOM 4 port PRI card. in the first port I have inserted 1 Live PRI.
And in the 2nd port, I have plugged RJ-45 cable ( crimped same as PRI cable ). The other end of this cable is then connected to the standard PBX. Now all the PRI calls need to be recorded. But its not happening.

I got the "circuit is busy : CONGESTION" Error.

All the time my standard PSTN PBX can not be used. So, I have made the other setup as well for testing.
============================================================================
I m having 1 Asterisk server with ATCOM 4 port PRI card.
Also have configured 2nd asterisk system (to work as PBX). This system is having 1 port PRI card.
I have inserted 1 Live PRI into the 1st Asterisk system (which is having 4 Port card.) In the second port I have plugged RJ-45 cable ( crimped same as PRI cable ). The other end of this cable is then connected to the 2nd asterisk system. (in 1 port card).
I want this second PC as PBX.
All the calls comes to the 1 live PRI, are forwarded to the second port in that system. and that second port should forward the call to the 2nd asterisk system. (as it acts as a PBX).

When any call comes to the PRI, it should dial out the other channel. but its not working.
its giving
"circuit is busy : CONGESTION" Error.

Please help me for this solution. Is there any other setup which I can apply. or any other solution for the PRI logger ?

In advance thanks for your support.

-Urmi

All the in coming and out going calls needs to be recorded.

[mazzic]

how about showing us some dialplan?

I've built a bridge box that took in calls on one PRI and routed them back out a PRI which was directly connected to the Merdian PBX.

You just build a context for the Incoming PRI that says to dial out the group the other PRI.

I would suspect that if you wish to record all of the calls you would just have to add a Monitor command prior to the dial.

Code: Select all

[from-pstn-pri]
exten => _X.,1,Monitor(,,b)
exten => _X.,n,Dial(DAHDI/g11/${EXTEN})


[urmi.l]

Thanks for your support. I know how to do the recording. But I can not connect the Standard PBX in parallel or I can not connect the other Asterisk system in parallel.

Below is my dialplan.

exten => _X.,1,Answer
exten=>_X.,,n,Set(CALLFILENAME=/var/spool/asterisk/monitor/${CALLERID(num)}-${STRFTIME(${EPOCH},%Y%m%d-%H%M)})
exten => _X.,n,Monitor(wav,${CALLFILENAME},m)
exten => _X.,n,Dial(DAHDI/g3/${EXTEN})

I want to consider the asterisk as PBX, And any calls on one PRI should be routed out a PRI which is directly connected to either to standard PBX or to other Asterisk (PBX).

My try : I have 4 port card in one PBX and 1 port card in other PBX. From the PRI I have taken the one parallel connection by crimping. and that parallel line I inserts into 1 port Card. But, then in both the asterisk the PRI goes down.
Does PRI not supports the parallel connection ? or Asterisk voip card doesnt supports this ? please guide me for this issue.

Thanks for your support.

-Urmi

-Urmi

[mazzic]

You need to make PRI cross over cables, which will then plug into the other asterisk box and your standard pbx. You also have to modify your DAHDI properly. The box with the live PRI have to be defined as net on the two connections, since it will be acting as the network side of those PRI connection.

t1 cross over cable http://www.t-radtalk.com/howto/how_to_make_a_t1_cross_over.htm

As a note, I just made little 1 inch cables and used a RJ-45 coupler to plug it into my system. That was I could turn any straight through cable into a cross over when needed.

[urmi.l]

Hello,

Following is my scenario :

1 standard (Panasonic) PBX : PRI is is connected to PBX.
1 Asterisk PBX with AX-4E card for call recording.

The PRI is splinted and I have taken one parallel connection from that. And that PRI parallel end is plugged into the AX-4E card. ( The parallel cable is RJ-45 cross cable as per your suggestion )

The PRI came up. The calls coming on the Standard PBX should also come to Asterisk. As the PRI line is parallel in Standard PBX and Asterisk. But there is no event on asterisk. I m not getting anything on asterisk.

extension.conf : I changed my dial plan as follows :

exten => _X.,1,Answer
exten=>_X.,n,Set(CALLFILENAME=/var/spool/asterisk/monitor/${CALLERID(num)}-${STRFTIME(${EPOCH},%Y%m%d-%H%M)})
exten => _X.,n,Monitor(wav,${CALLFILENAME},m)

For any incoming /outgoing calls from PRI are not coming to asterisk. even not getting any event on asterisk.I need the help.

Thanks for your support.

-Urmi

[mazzic]

Your scenario is confusing me.

The PRI is splinted and I have taken one parallel connection from that. And that PRI parallel end is plugged into the AX-4E card. ( The parallel cable is RJ-45 cross cable as per your suggestion )

The PRI came up. The calls coming on the Standard PBX should also come to Asterisk. As the PRI line is parallel in Standard PBX and Asterisk. But there is no event on asterisk. I m not getting anything on asterisk.

This leads me to believe that your trying to split the PRI. As in a Y cable connection. To my knowledge this isn't possible.

I think you need to bring the PRI into the Asterisk box. Then route the calls out a second PRI port on the Asterisk box to the Panasonic PBX.

The cross over cable would be needed from the Asterisk box to the Pasasonic PBX. On the Asterisk side of that connection it would be defined as NET since it is to the Pansonic the network side.

[david55]

You should probably not be using Answer(), as that starts charging.

[ianplain]

Hi

You need to bring the calls in and out of asterisk

http://cyber-cottage.co.uk/site/index.p ... &Itemid=60

we've done this many times, AFAIK you cannot monitor the isdn channels with a zapata card.

[urmi.l]

Hello,

Thank you very much for your reply.

Ok, I understood that the splited 'Y' cabling will not work. And change the scenario as per your suggestions.
I have first plugged the PRI in to the Asterisk Box and then cross over cable connected to from the Asterisk box to the Pasasonic PBX.

Will you please tell me how to route the calls out a second PRI port on the Asterisk box to the Panasonic PBX ???
Which command to use for this ??

Once again thanks for your support.

-Urmi

[mazzic]

Just use the dial command...

For instance if you set the context for the Telco PRI to PSTN-IN and group=1. Now define your other PRI (Asterisk to Panasonic PBX) as context PBX-IN and group=2.

Code: Select all

[PSTN-IN]
exten => _X.,1,Monitor(,,b)
exten => _X.,n,Dial(DAHDI/g2/${EXTEN})

[PBX-IN]
exten => _X.,1,Monitor(,,b)
exten => _X.,n,Dial(DAHDI/g1/${EXTEN})


so basically your just taking anything coming in on one PRI and sending it out the other.

[urmi.l]

Hello,

Thanks you very much for your support.
I have tried as per your suggestion and I can route the calls from one port to other. Also can connect to the standard PBX.
But the thing is the call first comes to one channel, and the recording is started. After that the call is routed to other channel. And there also the recording is started. The calls are working properly.

It generates 2 recording files for a single call. each is having single side voice.

Is 'MixMonitoring' be useful ??

-Urmi

[mazzic]

Actually it is not because the call is bridging that your getting two audio files, thats the way monitor works.

Yes, you can use mixmontor to join the two files into one...or you might try the m option flag on monitor.

from http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor

* flags: If flags contains the letter m, then when recording finishes, Asterisk will execute a unix program to combine the two sound files into a single sound file. By default, Asterisk will execute soxmix and then delete the original two sound files. Note that sox/soxmix may not necessarily understand the sound format (e.g. alaw) and can't therefore mix the in and out files down to one single file. You may specify a different mixing method by setting the MONITOR_EXEC channel variable to the path of the unix program you wish executed, then call Monitor to begin recording. At the completion of recording, the specified unix program will be executed with three command-line parameters: the two sound files and the filename where the program should save the combined sound file. In this situation, earlier versions of Asterisk will not delete the two original sound files; it's up to your program to do that if you need/wish to. The "m" flag is settable through the manager interface. Also see b - Don't begin recording unless a call is bridged to another channel.

[urmi.l]

Thanks you very much. Its working well now.

[urmi.l]

The PRI logger with Panasonic PBX is working fine. But there is one issue.

I make the Panasonic TDA200 pbx as QSIG-Master. And my asterisk box as slave (i.e pri_cpe). Everything is working fine. Extension wise recording is also working. The extensions are mapped with DID numbers. DID wise recording is also working.

But, my query is :
In outgoing calls, the DID should be displayed if the extensions are mapped with DID numbers (in pbx). But its not happening. For all the outgoing calls, it shows only the Pilot number of the PRI line. I have done all the possible changes in pbx. But still its shows pilot number only instead of DID numbers.

Do I need to do any configuration in asterisk ? If yes then please let me know.

-Urmi

[urmi.l]

Any help ???

[cxproin]

Hi Urmi.l

Just wanted to check if your PRI logger configuration on asterisk is working well?