Setup:
- Ibm 340 server with a.o. Digium TE405P quad T1/E1 card
- all 4 port in E1 mode
- Port 1 connected to an ISDN-30 PRI Telco trunk (Tele2) in pri_cpe
- Port 2 connected to an Alcatel 4200 E PBX Isdn-30 PRI in cpi_net
- Pbxinaflash, asterisk 1.4, Freepbx 2.40.0, centos 5.1
The Telco span is group 0, usually adressed by ZAP/g0 . The Pbx span is group 1, usually adresse by ZAP/g1
Up till now, the Telco PRI and the PBX RI where directly connected, and this has worked fine for serveral years
At this moment the * server is intended to act as a ‘router’ between the Telco PRI and the PBX PRI, and some Voip Sip telephones (Aastra).
- Connecting the incoming Telco calls to the PBX works fine
- Connecting PBX speeddial calls (which are passed as 1 string) to the Telco works fine
The problem is with manually dialed calls. The PBX tries to first connected to the other PRI, getting a dial tone, and then the digits are passed to the Telco
For Asterisk, this means the initially an empty dialnumber (extension) is passed. In the dialplan I can detect this by using the s extension, but since at that moment the actual dialnumber has not been entered yet, this gives problems for dialing the Telco.
I have tried the following
exten => s,1,Dial(ZAP/g0)
This failed
exten => s,1,Dial(ZAP/g0/0)
This failed, since the Telco wouldn’t accept a single 0
exten => s,1,SayDigits(1234)
This worked, but didn’t help connecting to the Telco
Then I tried
exten => s,1,DISA(no-password|internal)
This failed, with the PBX reporting that there was no ISDN device detected
When I preceeded the Disa with an Answer, DigitTimeOut(5) and/or ResponseTimeOut(10), this got the same result
For some reason, it looks like the Disa command doesn’t recognize it is used on a ISDN PRI Zap channel, and therefor should use the Proper ISDN protocol and/or act as an ISDN device.
How can I get asterisk let the PBX PRI connect to the Telco PRI and having the Telco PRI provide the Dialtone in case of an empty dialstring? The Disa functionality would be ideal, since it would enable Asterisk processing the entered dialnumber, but for the moment I would settle for a simple direct ‘bridge’, without the telco PRI assuming the full dialnumber was directly passes and completed already.
Please let me know if you need further information.
Best regards,
Peter Rens