Call Failed: 486 Busy Here


#1

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


#2

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 ?


#3

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


#4

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.


#5

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?


#6

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.


#7

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.


#8

[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.


#9

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


#10

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


#11

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


#12

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


#13

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


#14

Hello everybody,

Please help me regarding to this issue…

Thanks in advanced for your response.

Borin