Scenario 1: Place an analog call from a Uniden UIP200 phone connected to *@H1.5.
It takes approximately 10 seconds from the time the phone handset is lifted for the call to be dialed and audibly “picked up” at the other end.
From that point, there is a 20 second window before the call is dropped, and a “Fast Busy” is heard.
From the moment the call is audibly connected to the time that it is dropped, no “Call Timer” is visible on the LCD screen.
If you pick up the phone and dial nothing, a generated dialtone can be heard.
After 30 seconds, the PBX detects that no call has been placed and the “Fast Busy” is heard.
Obviously, no call has been placed and therefore there is no “Call Timer”.
If you pick up the phone and dial some PBX/Voicemail system. (Tested with “Fido” cellular here in Ontario).
Once the voicemail system kicks in, the voicemail greeting is heard and the “Call Timer” is not displayed UIP200 screen.
Once the voicemail greeting has ended, there is a “Beep/Tone” to signal the caller to begin leaving their message.
Magically, as soon as the “Beep/Tone” is heard the “Call Timer” begins to count.
As long as this occurs within 20 seconds of the call being “audibly connected” then the PBX “realizes” there is a call in progress.
When this happens, the call is not dropped and there is no “Fast Busy”.
If keys are pressed on the phone when the “Call Timer” is not visible, the DTMF tones are not heard and there is no remote response.
As soon as the “Call Timer” is displayed, the DTMF tones (that were apparently cached!) are played.
The initially thought was that the Uniden phones were somehow “timing out” and causing the issue.
I created a script on Asterisk that allowed me to call the PBX menu internally from the Uniden, and then type in phone number that was to be routed via analog.
Calls from the phone to the PBX menu internally are SIP calls; as a result, the “Call Timer” kicks in immediately as the PBX picks up and plays the menu.
After typing in the external number, the ringing was heard and the call was audibly “picked up”.
Exactly 30 seconds later, the call was dropped; I had configured Asterisk to play sound bytes to confirm the status of the call.
The wonderful voice of the PBX confirmed “No-one has answered your call”.
This proved that the Uniden phone was not “timing out”.
I then decided to focus on the magical “30 second” barrier by doing a GREP on the config files for Asterisk.
Again, nothing. No 30 references, so no 3000 millisecond references either.
Yes, set callprogress=yes. Yes, set progzone=ca.
Yes, set busydetect=yes.
Calls from the UIP200 over SIP (BroadVoice) work fine. It’s the ZAP channels giving me grief. Sometimes inbound, but usually outbound.
Rather than randomly pasting configs, any idea from the detail above? The system is vanilla *@H1.5 with no CVS updates. Yet.
UIP200 Firmware v4.63 if it should matter. TDM40B. ZTTOOL says no IRQs missed.