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Problems with calls through FXO trunk

8 years 6 months ago #10632 by juan.ing
Hello,

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.

Any ideas?

Thank you very much in advance.
Br,
Juan
8 years 6 months ago #10633 by Dario.hu
set the parameter busydetect=yes, busycount=6 at file /etc/asterisk/chan_dahdi.conf. if problem still ,add my skype: dario.hu2
8 years 6 months ago #10634 by juan.ing
Hi,

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?

Thanks
8 years 6 months ago #10635 by hua
8 years 6 months ago - 8 years 6 months ago #10636 by juan.ing
Hua,

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

Any idea?

Thanks,
Br,
Juan

hua wrote: Hi,

Please check this link www.openvox.cn/pub/manuals/Release/Engli...Quickstart_Guide.pdf

8 years 5 months ago #10642 by juan.ing
Hi Hua,

Any idea about this error code?

Thanks!!
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