I have a device based on the BF537 Stamp board. This device has one FXS module and one FXO module. I have been working for a while to get the Zaptel drivers working and that is finally going everything is working quite nicely except for one aspect:
For outbound calls (I have a local phone connected to the FXS board and the Exchange connected to the FXO board) I have immediate=yes so it acts as a hot phone and auto dials extension 308 on our office exchange. This works perfectly and the 308 extension starts ringing - I thought I was done however I found that the asterisk CLI shows:
– Called g2/308
– Zap/2-1 answered Zap/1-1
– Native bridging Zap/1-1 and Zap/2-1
The Zap/2-1 answers the call and bridges the FXO/FXS device while the 308 Exchange phone is still ringing. I assume this is needed so the local phone can be bridged to the office exchange.
However this does create 2 problems for me:
My cdr call files are incorrect since all outbound calls show as answered after 2 seconds even if the phone on the exchange never answers the call
I have a originate script that auto dials the exchange or local phone, and then plays a automated message when answered. This is needed for testing. On the local phone it works fine however on the phone connected to the Exchange I get the message half way through. This is obviously caused since Asterisk sees the call as answered (when the FXO (ZAP/2) device answers) and then starts playing the message.
Zaptel has been replaced by dahdi. As far as I know, no currently supported versions of Asterisk still support zaptel.
Does your PSTN provider provide answer supervision? If so how do they signal it? If not, there is no way for Asterisk to know when the call is answered, so it must assume it is answered immediately.
If they do provide a supported form of answer supervision (unlikely if the line was not intended for PABX use) you will need to modify chan_dahdi.conf (assuming you upgrade to supported software) to match the supervision method they use. (zaptel.conf probably can do this as well.)
Thanks for the reply. I am trying to think of a work around for my problems so I just want to get some feedback since I am not very clued up on everything asterisk related.
The system is being used on an existing Exchange where we will patch in between the phones and the exchange. Because of this we will be able to capture the CDR info on both phones and therefore I will be able to record only the incoming calls where the data is accurate and use this only. The phones are also going to be hot phones therefore I will always know where the call came from.
On my second problem I am not sure how to solve this other than maybe having a message on a loop. I.E. instead of hearing half the message when I automate a call the message just loops the whole time using a playback and a goto. This is really the only thing I can think of is there anything else I can consider doing here instead?