I am trying to stablish calls to the PSTN through openvox analog trunk. I have configured FXO port with Spanish region tone and TBR21 with fxorxgain=6, fxotxgain=6.
Outgoing calls progress ok but something is wrong because the disconnection does not finish properly. Once hunged up, the line is still busy. After a certain timeout, the operator proceeds with the disconnection. Until that point no other call can be made. Any problem with the disconnection tone in spanish config? I have also tried with the french config, facing the same problem.
Thanks for your response. It is working fine now.
Now, I am having problems with incomming calls. I have configured an incomming rule. From trunk openvox to an internal SIP extension. The calls are not redirected to the sip phone. Any ideas?
Yes, I followed that quickstart guide and just changed Destination to User extension 6005 instead of 6088, the default FXS ext.
This is working fine when I configure incoming calling rule from an Openvox GSM gateway trunk to the same internal sip extension (6005).
I get this error in asterisk messages log file when the call is entering:
chan_dadhi.c: DTMFCID timed out waiting for ring. Exinting simple switch