# Call Failed: 486 Busy Here **URL:** \ **Category:** Asterisk Support\ **Created:** [March 14, 2006, 7:33am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011 "2006-03-14T07:33:15Z")\ **Posts on this page:** 14\ **Page:** 1
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 7:33am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/1 "2006-03-14T07:33:15Z")
Dear All I install the Asterisk and I´m doing some tests with X-lite, I configured 2 extension, and try to call each other, but in both aways the X-lite always says Call Failed 486 Busy Here,but the extensions are not busy. When I enable the Voicemail & Directory, I get “The personal extention” and Hung Up. My Asterisk version 1.2.4 and CentOS release 4.2 (Final) How can I fix it? Looking for your kind response. Best regards, Borin ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 14, 2006, 9:01am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/2 "2006-03-14T09:01:17Z")
sounds more like you’re using A@H ?? the “no-digits” thing is a permissions problem … look at [forums.digium.com/viewtopic.php? … ght=#15679](http://forums.digium.com/viewtopic.php?p=15679&highlight=#15679) can you call into the asterisk box from the individual phones ? ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 9:51am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/3 "2006-03-14T09:51:59Z")
baconbuttie: Thanks for your reply. Yes, of course I am using A@H, After I do the chmod 755 /var/lib/asterisk/sounds/digits 1 - If I did not enable the voice mail in each extention, I still get the error “Call Failed 486 Busy Here” the same as before 2 - If I enable the voice mail in each extention, A - 201, B - 202 , both of account was properly config in Xlite and register with A@H. When I call I get the message “The personal Extention, ---- was no online, please leave a message after the tone”…why? I think that extention 201 can talk with extention 202 at different PC. Maybe I can not call into the asterisk box from the individual phones. I am not sure. Please advice how to fix this problem. Looking for your kind advice. Best regards, Borin ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 14, 2006, 10:07am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/4 "2006-03-14T10:07:23Z")
i’m confused. do you have both accounts setup on a single XLite installation on a single PC ? to be honest, i’ve never seen an A@H install, with softphones setup on separate PCs, ever fail out-of-the-box. you must have either configured your extensions incorrectly, or you’ve got your client setup wrong. ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 10:17am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/5 "2006-03-14T10:17:03Z")
This is my senario: One Server run A@H PC 01 - Xlite config with extention 201 PC 02 - Xlite config with extention 202 I want to call from ext. 201 to ext 202. So that people at ext. 201 can talk with people at ext. 202. What do you think? ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 14, 2006, 10:31am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/6 "2006-03-14T10:31:18Z")
do you see your UAs register correctly with the Asterisk box ? have you tried using each UA individually to connect to a server-side extension (e.g. \*43) ? is DND or call-forwarding turned off for both extensions ? does the same thing happen regardless of call direction ? these are all fairly simple fault-finding steps … lower the number of variables to help you determine where the problem is. ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 10:47am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/7 "2006-03-14T10:47:53Z")
Yes, I see it register in A@H box. I can connect to server with ext. \*43 to do the echo testing. I am not sure where is DND or call-forwarding? please give me more detail about this. lower the number of variables refer to A@H or other config? Thanks. ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 14, 2006, 10:53am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/8 "2006-03-14T10:53:33Z")
[quote=“Borin”]please give me more detail about this.[/quote]no, why don’t you go and read the A@H documentation about call-forwarding and how to activate/de-activate it. ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 11:06am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/9 "2006-03-14T11:06:03Z")
From my Xlite I dial to \*73 to disable my call forwarding at both extention mean at PC 1 and PC 2. But I still get the same problem “Call Failed 486 Busy Here” Any more advice? Thanks ---
**Author:** ![baconbuttie](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/baconbuttie/32/916_2.png) [@baconbuttie](https://community.asterisk.org/u/baconbuttie)\ **Post date:** [March 14, 2006, 11:54am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/10 "2006-03-14T11:54:03Z")
turn up the verbosity of your log and post it here when you’ve made a failed call. ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 14, 2006, 2:17pm UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/11 "2006-03-14T14:17:31Z")
The following is my log: Feb 3 10:58:17 VERBOSE[2623] logger.c: – Registered SIP ‘202’ at 192.168.0.106 port 5060 expires 1800 Feb 3 10:58:24 DEBUG[2623] chan\_sip.c: Stopping retransmission on ‘07ee29f81e1e742b11d9f38138f20372@192.168.0.105’ of Request 102: Match Found Feb 3 10:58:32 DEBUG[2623] chan\_sip.c: Auto destroying call '7AE1FAF173274E6BA431FD554985BFB8@192.168.0.105’ Feb 3 10:58:55 DEBUG[2623] chan\_sip.c: Setting NAT on RTP to 0 Feb 3 10:58:55 DEBUG[2623] chan\_sip.c: Stopping retransmission on ‘733E58B6-D3F7-4787-821D-18FB9D1B842C@192.168.0.222’ of Response 36820: Match Found Feb 3 10:58:55 DEBUG[2623] chan\_sip.c: Setting NAT on RTP to 0 Feb 3 10:58:55 DEBUG[2623] chan\_sip.c: Checking SIP call limits for device 201 Feb 3 10:58:55 DEBUG[2623] chan\_sip.c: build\_route: Contact hop: Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing Macro(“SIP/201-8760”, “exten-vm|novm|202”) in new stack Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing Macro(“SIP/201-8760”, “user-callerid”) in new stack Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing DBget(“SIP/201-8760”, “AMPUSER=DEVICE/201/user”) in new stack Feb 3 10:58:55 VERBOSE[3387] logger.c: – DBget: varname=AMPUSER, family=DEVICE, key=201/user Feb 3 10:58:55 VERBOSE[3387] logger.c: – DBget: set variable AMPUSER to 201 Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing DBget(“SIP/201-8760”, “AMPUSERCIDNAME=AMPUSER/201/cidname”) in new stack Feb 3 10:58:55 VERBOSE[3387] logger.c: – DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=201/cidname Feb 3 10:58:55 VERBOSE[3387] logger.c: – DBget: set variable AMPUSERCIDNAME to Borin Feb 3 10:58:55 DEBUG[3387] pbx.c: Expression result is '0’ Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “0?5”) in new stack Feb 3 10:58:55 DEBUG[3387] pbx.c: Not taking any branch Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing SetCallerID(“SIP/201-8760”, ““Borin” \<201\>”) in new stack Feb 3 10:58:55 VERBOSE[3387] logger.c: – Executing NoOp(“SIP/201-8760”, “Using CallerID “Borin” \<201\>”) in new stack Feb 3 10:58:56 VERBOSE[3387] logger.c: – Executing SetVar(“SIP/201-8760”, “FROMCONTEXT=exten-vm”) in new stack Feb 3 10:58:56 VERBOSE[3387] logger.c: – Executing Macro(“SIP/201-8760”, “record-enable|202|IN”) in new stack Feb 3 10:58:56 DEBUG[3387] pbx.c: Function result is '0’ Feb 3 10:58:56 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “0 \> 0?2:4”) in new stack Feb 3 10:58:56 VERBOSE[3387] logger.c: – Goto (macro-record-enable,s,4) Feb 3 10:58:56 VERBOSE[3387] logger.c: – Executing AGI(“SIP/201-8760”, “recordingcheck|20000203-105856|949550335.15”) in new stack Feb 3 10:58:56 VERBOSE[3387] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck Feb 3 10:58:58 VERBOSE[3387] logger.c: recordingcheck|20000203-105856|949550335.15: Inbound recording not enabled Feb 3 10:58:58 VERBOSE[3387] logger.c: – AGI Script recordingcheck completed, returning 0 Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing NoOp(“SIP/201-8760”, “No recording needed”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing Macro(“SIP/201-8760”, “dial|15|tr|202”) in new stack Feb 3 10:58:58 DEBUG[3387] pbx.c: Expression result is '0’ Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “0?4:2”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Goto (macro-dial,s,2) Feb 3 10:58:58 DEBUG[3387] pbx.c: Function result is ‘0’ Feb 3 10:58:58 DEBUG[3387] pbx.c: Expression result is ‘0’ Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “0?5:4”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Goto (macro-dial,s,4) Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing AGI(“SIP/201-8760”, “dialparties.agi”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi Feb 3 10:58:58 VERBOSE[3387] logger.c: – AGI Script dialparties.agi completed, returning 0 Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing NoOp(“SIP/201-8760”, “Returned from dialparties with no extensions to call”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing SetVar(“SIP/201-8760”, “DIALSTATUS=BUSY”) in new stack Feb 3 10:58:58 WARNING[3387] ast\_expr2.y: non-numeric argument Feb 3 10:58:58 DEBUG[3387] pbx.c: Expression result is ‘0’ Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “0?s-BUSY|1”) in new stack Feb 3 10:58:58 DEBUG[3387] pbx.c: Not taking any branch Feb 3 10:58:58 DEBUG[3387] pbx.c: Expression result is ‘1’ Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing GotoIf(“SIP/201-8760”, “1?s-BUSY|1”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Goto (macro-exten-vm,s-BUSY,1) Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing NoOp(“SIP/201-8760”, “Extension is reporting BUSY and has no Voicemail”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing Busy(“SIP/201-8760”, “”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on ‘SIP/201-8760’ in macro ‘exten-vm’ Feb 3 10:58:58 VERBOSE[3387] logger.c: == Spawn extension (from-internal, 202, 1) exited non-zero on ‘SIP/201-8760’ Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing Macro(“SIP/201-8760”, “hangupcall”) in new stack Feb 3 10:58:58 DEBUG[2623] chan\_sip.c: Stopping retransmission on ‘733E58B6-D3F7-4787-821D-18FB9D1B842C@192.168.0.222’ of Response 36821: Match Not Found Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing ResetCDR(“SIP/201-8760”, “w”) in new stack Feb 3 10:58:58 DEBUG[3387] cdr\_addon\_mysql.c: cdr\_mysql: inserting a CDR record. Feb 3 10:58:58 DEBUG[3387] cdr\_addon\_mysql.c: cdr\_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2000-02-03 10:58:55’,’“Borin” \<201\>’,‘201’,‘202’,‘from-internal’, ‘SIP/201-8760’,’’,‘ResetCDR’,‘w’,3,0,‘NO ANSWER’,3,’’,‘949550335.15’) Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing NoCDR(“SIP/201-8760”, “”) in new stack Feb 3 10:58:58 WARNING[3387] cdr.c: CDR on channel ‘SIP/201-8760’ not posted Feb 3 10:58:58 WARNING[3387] cdr.c: CDR on channel ‘SIP/201-8760’ lacks end Feb 3 10:58:58 VERBOSE[3387] logger.c: – Executing Wait(“SIP/201-8760”, “5”) in new stack Feb 3 10:58:58 VERBOSE[3387] logger.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on ‘SIP/201-8760’ in macro 'hangupcall’ Feb 3 10:58:58 VERBOSE[3387] logger.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-8760’ Feb 3 10:58:58 DEBUG[3387] chan\_sip.c: update\_call\_counter(201) - decrement call limit counter Feb 3 10:58:58 DEBUG[3387] chan\_sip.c: update\_call\_counter(201) - decrement call limit counter Looking for your kind advice. Borin ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 16, 2006, 1:01am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/12 "2006-03-16T01:01:13Z")
Anybody know how to fix this problem? I am looking all forum regarding to this problem but nobody post the working solution at all. Your commends and advices will be most welcome. Thanks in advanced. Borin ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 21, 2006, 1:42am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/13 "2006-03-21T01:42:06Z")
I just do the new installation with AAH 2.7 and I still get the same error problem. I really need help regarding to this problem. Please help me… Your response will be highly appriciated. Borin ---
**Author:** ![Borin](https://avatars.discourse-cdn.com/v4/letter/b/a6a055/32.png) [@Borin](https://community.asterisk.org/u/Borin)\ **Post date:** [March 24, 2006, 1:30am UTC](https://community.asterisk.org/t/call-failed-486-busy-here/5011/14 "2006-03-24T01:30:19Z")
Hello everybody, Please help me regarding to this issue… Thanks in advanced for your response. Borin