Call Failed: 486 Busy Here

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

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

can you call into the asterisk box from the individual phones ?

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

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.

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?

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.

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.

[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.

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

turn up the verbosity of your log and post it here when you’ve made a failed call.

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

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

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

Hello everybody,

Please help me regarding to this issue…

Thanks in advanced for your response.

Borin