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Summary:ASTERISK-09202: Call Drops after 5 to 20 minutes
Reporter:Phone User (phone008)Labels:
Date Opened:2007-04-05 12:55:59Date Closed:2007-08-21 15:49:12
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 070405_Drop.txt
Description:I am having constant problems with dropped analog calls on my Polycom 601 (connected to PSTN via a Digium TDM card).  The problem occurs regardless of whether I call out or receive a call.  Everything is great, then I just receive a drop after anywhere from 5 to 20 minutes.

I reviewed a similar (but closed) reported bug and turned on all the debugging goodies.  A copy of the file is attached (070405_Drop.txt).  The history of the call in question is as follows:  I receved a call from an outside PSTN caller at around 9:43:51 a.m.  I believe the call number is 7f7d8ff66abb736334d1dd972a5aadc3.  The call suddenly drops at around 10:03:37.  

Thank you for all your hard work.

****** ADDITIONAL INFORMATION ******

Setup:
Digium TDM02B with 2 FXO ports connected to PSTN lines
Trixbox 2.0 (clean install with TDM card inserted before Trixbox installation)
Polycom 601 - main phone
Polycom 501 x2 (installed but never used)
Comments:By: Tilghman Lesher (tilghman) 2007-04-05 13:44:29

Please upgrade your Polycom phone to the latest firmware.  There are multiple problems with the version 2.0 firmware, which should be solved in 2.1.
By: Phone User (phone008) 2007-04-05 18:47:18

I upgraded to sip 2.1.0.2708 and will let you know of the results.  Thanks.
By: Phone User (phone008) 2007-04-06 17:03:07

Upgrading the firmware did not help.  Someone called me from the PSTN and the call dropped after 6 min 46 seconds.  I was hopeful that the problem had been resoved, because earlier in the day I had a successful 37 minute call.  Do you need me to attach another log, or can you work with the one I've given you?
By: Serge Vecher (serge-v) 2007-04-09 09:40:20

The following precedes the call drop.
Apr  5 10:04:26 DEBUG[9548] chan_zap.c: Exception on 17, channel 4
Apr  5 10:04:26 DEBUG[9548] chan_zap.c: Got event On hook(1) on channel 4 (index 0)
Apr  5 10:04:26 DEBUG[9548] chan_zap.c: disabled echo cancellation on channel 4
Apr  5 10:04:26 DEBUG[9548] channel.c: Didn't get a frame from channel: Zap/4-1
Apr  5 10:04:26 DEBUG[9548] channel.c: Bridge stops bridging channels Zap/4-1 and SIP/101-082cf8d0

Since the exception is generated within the WC, the best resource to deal with such an issue is support@digium.com . Please contact support and reopen the issue if they direct you so. Thanks.
By: John Bigelow (jbigelow) 2007-08-21 15:49:12

For anyone that runs across this "bug"(not really a bug, just needed tweaked due to line conditions), this issue was that the voltage on the POTS line was dropping some during active calls. The voltage drop hit the zaptel drivers default thresh hold and zaptel considered it a hang up coming from the telco. The drivers(wctdm, wctdm24xxp) can be modified to make them more tolerant of voltage drops by lowering the DEFAULT_BATT_THRESH define in the driver. This can also be passed as an argument while loading the driver using "battthresh=x". Replace the "x" with the value you want to set. The default is 3. In this case lowering it to a value of 2 solved the problem.

-Digium support