[Asterisk-Users] RE: Dropping Calls, irregular interval no logs

[Jared Armstrong]

Ok, I finally got asterisk to output the 'full' log so I could review it
and it finally dropped a call on me. As you can see in the log below,
asterisk thinks it is seeing the busy signal during a call that has been
in progress for over 30 seconds, I have set busydetect=no in my
Zapata.conf file, but wonder if anyone else agrees with this. This is
the output I see when it dropped the call:

 

2004-12-08 15:19:19 DEBUG[4650]: Device 'SIP/136' changed to state '2'

2004-12-08 15:19:23 DEBUG[4650]: Acked pending invite 102

2004-12-08 15:19:23 DEBUG[4650]: Stopping retransmission on
'0ab2e6632b48cdaf7479ce5d7d277c30 at 192.168.4.25' of Request 102: Found

2004-12-08 15:19:23 DEBUG[4650]: Oooh, we need to change our formats
since our peer supports only 0x4(ULAW) and not 0x2(GSM)

2004-12-08 15:19:23 DEBUG[4650]: build_route: Contact hop:
<sip:136 at 192.168.4.85>

2004-12-08 15:19:23 VERBOSE[4650]:     -- SIP/136-8aa2 answered Zap/1-1

2004-12-08 15:19:23 DEBUG[4650]: Requested indication -1 on channel
Zap/1-1

2004-12-08 15:19:23 DEBUG[4650]: Ooh, format changed from UNKN to ULAW

2004-12-08 15:20:53 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:07 DEBUG[4650]: Device 'SIP/135' changed to state '0'

2004-12-08 15:21:18 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:18 DEBUG[4650]: Stopping retransmission on
'66d2b161133ebf6810f91ec16e5081da at 192.168.4.25' of Request 102: Found

2004-12-08 15:21:22 DEBUG[4650]: Auto destroying call
'3812ff91-7ea5ee5b-8188aafe at 192.168.4.86'

2004-12-08 15:21:30 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:30 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:36 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:36 DEBUG[4650]: Setting NAT on RTP to 0

2004-12-08 15:21:50 DEBUG[4650]: Requesting Hangup because the busy tone
was detected on channel Zap/1-1

2004-12-08 15:21:50 DEBUG[4650]: Didn't get a frame from channel:
Zap/1-1

2004-12-08 15:21:50 DEBUG[4650]: Bridge stops bridging channels Zap/1-1
and SIP/136-8aa2

2004-12-08 15:21:50 DEBUG[4650]: update_user_counter(136) - decrement
outUse counter

2004-12-08 15:21:50 DEBUG[4650]: Exiting with DIALSTATUS=ANSWER.

2004-12-08 15:21:50 VERBOSE[4650]:   == Spawn extension (macro-stdexten,
s, 4) exited non-zero on 'Zap/1-1' in macro 'stdexten'

2004-12-08 15:21:50 VERBOSE[4650]:   == Spawn extension (office-day,
136, 6) exited non-zero on 'Zap/1-1'

2004-12-08 15:21:50 DEBUG[4650]: Device 'SIP/136' changed to state '0'

2004-12-08 15:21:50 DEBUG[4650]: Hangup: channel: 1 index = 0, normal =
16, callwait = -1, thirdcall = -1

2004-12-08 15:21:50 DEBUG[4650]: disabled echo cancellation on channel 1

2004-12-08 15:21:50 DEBUG[4650]: Set option TDD MODE, value: OFF(0) on
Zap/1-1

 

Thanks,

 

Jared Armstrong

  _____  

From: Jared Armstrong 
Sent: Wednesday, December 08, 2004 1:12 PM
To: 'asterisk-users at lists.digium.com'
Subject: RE: Dropping Calls, irregular interval no logs

 

Ok, So I removed the Qualify=2000 from my sip.conf file and set NAT=no
and I changed the connection negotiation in my Polycom config to 3600
(default) and now I haven't had a dropped call yet. Does anyone know
which of these if any might be the cause or if there is something else I
am still overlooking?

 

Jared Armstrong

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