I installed asterisk 1.2.22, with driver zaptel 1.4.6 and a digium tdm400p with 4 fxo modules.
It always happens that when i call a number on the pstn, and the other end hangs up, asterisk doesnât detect it and continues as if the call is still running. It doesnât happen if i hangup before with my sip phone.
i canât understand why the chan_zap.so module believes that the digium card is for isdn (in other computers it doensât happen)
When the pstn number hangs up, the logfile is full of this lines:
Nov 18 15:13:34 DEBUG[3156] dsp.c: ast_dsp_busydetect detected busy, avgtone: 220, avgsilence 180
Nov 18 15:13:34 DEBUG[3156] dsp.c: Requesting Hangup because the busy tone was detected on channel Zap/4-1
but it doensât close the call, it doensât stop bridging the zap channel with the sip channel created for the sip phone.
Iâm sure it is a ridiculous problem but i canât understand where i wrong. I add hereby the zapata.conf hoping that it can help
Hi Dovid
I have the same versions of asterisk and zaptel, with the same digium card, on other two different computers and it works.
Could it be something else?
By the way there arenât any differences between programsâ versions and asterisk configuration in all of the three computers i worked on.
From what I have seen in the past you need to use the same versions of asterisk and zaptel. You have me stumped with the issue. I am not the zatepl expert. Try speaking to tzafrir on the #asterisk channel on irc (irc.freenode.net). He is very knowledgeable when it comes to zaptel.