I have a new Digium AEC800 with 2 4-port FXO cards and the Echo cancel daughter card installed with the latest PiaF download Asterisk 1.4 32-bit and Orgasm52 loaded.
I found the thread "ZAP TRUNK problem after update-scrips / update fixes ???"
and applied it by changing the from-pstn to from-zaptel.
I had it working yesterday.
I rebooted the server this morning and it will not longer recognize the touch tone entry to go to an extension.
If I call from the Cisco phone to the Aastra phone all is fine.
If I call out, it dials the outside extension/number just fine but there is not audio from the pstn user back to me.
I have applied the one-way audio patch earlier - remove the comments in the file ... but that should not cause the problem as the LAN phones work just fine.
So I have not audio if I call into the system on one of the PSTN lines not only is there no inbound audio to the IVR default routed telephone on the LAN but the touchtones to select an extension are also of no affect.
Also, there is no Caller-ID passed after the reboot.
This is the last problem with the setup so that I can get on to the work at hand.
The .call creation code, etc. is already written and tested on an old Dell P4 with 2 X100P cards and it works great.
I just cannot get the outside interaction with the PSTN to work.
It was working perfectly before the reboot - go figure!
The system is a Dell 1850 with the PCIe riser installed and the Digium AEX800 card installed.
There a 4 analog lines from our Avaya PBX attached to the card.
The /etc/asterisk/zapata-channels.conf still has the from-zaptel for all the lines.
I have run amportal restart and the values still stay the same - as long as I don't do genzaptelconf, then I have to update the file again.
This one is really a problem!
I guess the moral of the story is to NEVER reboot!!!
Any assistance is GREATLY appreciated!
Thanks
TomS
I found the thread "ZAP TRUNK problem after update-scrips / update fixes ???"
and applied it by changing the from-pstn to from-zaptel.
I had it working yesterday.
I rebooted the server this morning and it will not longer recognize the touch tone entry to go to an extension.
If I call from the Cisco phone to the Aastra phone all is fine.
If I call out, it dials the outside extension/number just fine but there is not audio from the pstn user back to me.
I have applied the one-way audio patch earlier - remove the comments in the file ... but that should not cause the problem as the LAN phones work just fine.
So I have not audio if I call into the system on one of the PSTN lines not only is there no inbound audio to the IVR default routed telephone on the LAN but the touchtones to select an extension are also of no affect.
Also, there is no Caller-ID passed after the reboot.
This is the last problem with the setup so that I can get on to the work at hand.
The .call creation code, etc. is already written and tested on an old Dell P4 with 2 X100P cards and it works great.
I just cannot get the outside interaction with the PSTN to work.
It was working perfectly before the reboot - go figure!
The system is a Dell 1850 with the PCIe riser installed and the Digium AEX800 card installed.
There a 4 analog lines from our Avaya PBX attached to the card.
The /etc/asterisk/zapata-channels.conf still has the from-zaptel for all the lines.
I have run amportal restart and the values still stay the same - as long as I don't do genzaptelconf, then I have to update the file again.
This one is really a problem!
I guess the moral of the story is to NEVER reboot!!!
Any assistance is GREATLY appreciated!
Thanks
TomS
Phone System 