How to disable call queues..?

The output tells you what is wrong, I know its easy to keep asking other people what is going wrong.
But as you are a beginner, you need to read and analyse first, then if you still cant see what is going wrong then ask a question…

[Sep 27 15:55:26] WARNING[47052][C-00000007]: pbx_functions.c:460 func_args: Can’t find trailing paren thesis for function ‘GROUP_COUNT($EXTEN’?

What would “Cant find trailing parenthesis for funciont” mean…?

From your configuration:

same => n,ExecIf($[${GROUP_COUNT($EXTEN}@user1)} > 1]?Busy());

If I were you, I would check to see if all the (, { and [ match. When you can answer that… You will have found the cause of the error.

@meightee Thanks to the guidance…It was my mistake.sorry…i had corrected the script…now no more error …but both calls are landing…no busy tone for second caller like we intended.

Saved useragent “SIPPER for PhonerLite” for peer user1
== Using SIP RTP CoS mark 5
0x7fe574009250 – Strict RTP learning after remote address set to: 10.0.xx.xx:29100
– Executing [1544@tutorial:1] NoOp(“SIP/asterisk-00000000”, “”) in new stack
– Executing [1544@tutorial:2] Set(“SIP/asterisk-00000000”, “GROUP(user1)=1544”) in new stack
– Executing [1544@tutorial:3] ExecIf(“SIP/asterisk-00000000”, “0?Busy()”) in new stack
– Executing [1544@tutorial:4] Dial(“SIP/asterisk-00000000”, “SIP/user1”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/user1
– SIP/user1-00000001 is ringing
0x7fe5a8006660 – Strict RTP learning after remote address set to: 192.168.xx.xx:5062
– SIP/user1-00000001 answered SIP/asterisk-00000000
– Channel SIP/user1-00000001 joined ‘simple_bridge’ basic-bridge
– Channel SIP/asterisk-00000000 joined ‘simple_bridge’ basic-bridge
0x7fe5a8006660 – Strict RTP switching to RTP target address 192.168.xx.xx:5062 as source
0x7fe574009250 – Strict RTP switching to RTP target address 10.0.xx.xx:29100 as source
0x7fe574009250 – Strict RTP learning complete - Locking on source address 10.0.xx.xx:29100
0x7fe5a8006660 – Strict RTP learning complete - Locking on source address 192.168.xx.xx:5062
== Using SIP RTP CoS mark 5
0x7fe5740150e0 – Strict RTP learning after remote address set to: 10.0.xx.xx:29104
– Executing [1544@tutorial:1] NoOp(“SIP/asterisk-00000002”, “”) in new stack
– Executing [1544@tutorial:2] Set(“SIP/asterisk-00000002”, “GROUP(user1)=1544”) in new stack
– Executing [1544@tutorial:3] ExecIf(“SIP/asterisk-00000002”, “0?Busy()”) in new stack
– Executing [1544@tutorial:4] Dial(“SIP/asterisk-00000002”, “SIP/user1”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/user1
– SIP/user1-00000003 is ringing
== Spawn extension (tutorial, 1544, 4) exited non-zero on ‘SIP/asterisk-00000002’
– Channel SIP/asterisk-00000000 left ‘simple_bridge’ basic-bridge
– Channel SIP/user1-00000001 left ‘simple_bridge’ basic-bridge
== Spawn extension (tutorial, 1544, 4) exited non-zero on ‘SIP/asterisk-00000000’

Can any one guide me with your expertness?..I m stuck over here…I had tried by adjusting various busy parameters still no luck.