# Rejecting calls with no CLI **URL:** \ **Category:** Asterisk Support\ **Created:** [October 13, 2011, 11:03am UTC](https://community.asterisk.org/t/rejecting-calls-with-no-cli/36999 "2011-10-13T11:03:13Z")\ **Posts on this page:** 2\ **Page:** 1
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [October 13, 2011, 11:03am UTC](https://community.asterisk.org/t/rejecting-calls-with-no-cli/36999/1 "2011-10-13T11:03:13Z")
Supporting Asterisk is not my day-to-day job, so apologies to everyone if this reported elsewhere. Whenever an incoming call is made to Asterisk with the callers CLI restricted, then it is rejecting the call with a 603. Original Invite message INVITE sip:+447924571xxx@194.54.xxx.xxx:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 194.54.xxx.x:5060;branch=z9hG4bK38951383 From: sip:anonymous@anonymous.invalid;tag=1446077424 To: sip:+447924571xxx@194.54.xxx.xxx:5060;user=phone Call-ID: [249245709@194.54.xxx.x](mailto:249245709@194.54.xxx.x) CSeq: 20 INVITE Contact: sip:COSPEED@194.54.xxx.x SIP/2.0 603 Declined Via: SIP/2.0/UDP 194.54.xxx.x:5060;branch=z9hG4bK38951383 From: sip:anonymous@anonymous.invalid;tag=1446077424 To: sip:+447924571xxx@194.54.xxx.xxx:5060;user=phone;tag=as2c482140 Call-ID: [249245709@194.54.xxx.x](mailto:249245709@194.54.xxx.x) CSeq: 20 INVITE Server: Asterisk PBX SVN-branch-1.8-r310088M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Reason: Q.850;cause=21 Content-Length: 0 Is there some dial plan setting somewhere that checks for this? ---
**Author:** ![david55](https://avatars.discourse-cdn.com/v4/letter/d/e19adc/32.png) [@david55](https://community.asterisk.org/u/david55)\ **Post date:** [October 13, 2011, 11:20am UTC](https://community.asterisk.org/t/rejecting-calls-with-no-cli/36999/2 "2011-10-13T11:20:29Z")
That example does have a caller ID; it is “anonymous”. If there were no caller ID, there would be no @ in the From header. Although there are sample dialplans, Asterisk’s initial dialplan is empty, so you, or your GUI writer, would have to actively add something to reject a call from the dialplan. I suspect this may actually be a failure to match sip.conf.