# Dialplan **URL:** \ **Category:** Asterisk Support\ **Created:** [January 29, 2016, 9:34pm UTC](https://community.asterisk.org/t/dialplan/65010 "2016-01-29T21:34:49Z")\ **Posts on this page:** 1\ **Page:** 1
**Author:** ![geraldinejns](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/geraldinejns/32/55_2.png) [@geraldinejns](https://community.asterisk.org/u/geraldinejns)\ **Post date:** [January 29, 2016, 9:34pm UTC](https://community.asterisk.org/t/dialplan/65010/1 "2016-01-29T21:34:49Z")
What am i doing wrong? I created a RAW and EZ, dialplan. You can’t get no basic than this. When i click the Dial button on the XLite softphone and dial 201 the XLite phone rings,and rings, and rings until i hit the end but on the Xlite softphone. The same thing happens on my polycom soundpoint 330. When i dial 101 the polycom phone rings, and rings, and rings until i pick up the handset and slam it down 😃. Below is my easiest configuration. I want to learn how to do this from scratch. Below is the sip.conf [general] context=public ; Default context for incoming calls. Defaults to 'default’ allowoverlap=no ; Disable overlap dialing support. (Default is yes) udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all) tcpenable=no ; Enable server for incoming TCP connections (default is no) tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces) transport=udp ; Set the default transports. The order determines the primary default transport. srvlookup=yes qualify=yes ; Enable DNS SRV lookups on outbound calls [authentication] basic-options ; a template dtmfmode=rfc2833 context=from-office type=friend natted-phone ; another template inheriting basic-options directmedia=no host=dynamic public-phone ; another template inheriting basic-options directmedia=yes my-codecs ; a template for my preferred codecs disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw ulaw-phone ; and another one for ulaw-only disallow=all allow=ulaw [101] type=friend context=phones host=dynamic secret=asterisk allow=ulaw ## [201] type=friend context=phones host=dynamic secret=asterisk allow=ulaw Below is Extensions.conf [phones] exten =\> 101,1,NoOp(first line) exten =\> 101,2,NoOp(second line) exten =\> 101,3,Dial(SIP/201) exten =\> 101,4,Hangup exten =\> 201,1,NoOp(first line) exten =\> 201,2,NoOp(second line) exten =\> 201,3,Dial(SIP/101) exten =\> 201,4,Hangup * * * Output from sip show peers ## asterisksystem_CLI\> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description 101/101 10.0.0.2 D a 52848 OK (107 ms) 201/201 10.0.0.4 D a 5060 OK (14 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] asterisksystem_CLI\> Below is the out put when i dial from my polycom soundpoint 330 phone which is extension 201 to xlite extension 101. When i dial extension 201 from the xlite the polycom phone does not ring, and vice-versa. But it shows on the output below. == Using SIP RTP CoS mark 5 – Executing [101@phones:1] NoOp(“SIP/201-00000004”, “call for line two”) in new stack – Executing [101@phones:2] Dial(“SIP/201-00000004”, “SIP/201”) in new stack == Using SIP RTP CoS mark 5 – Called SIP/201 – SIP/201-00000005 is ringing == Spawn extension (phones, 101, 2) exited non-zero on 'SIP/201-00000004’ asterisksystem\*CLI\>