Asterisk did not work if configure from freepbx

General Help
1

I’ve install asterisk (latest version). configure trough *.conf and works fine.
Then i install freepbx, installation runs fine and i could configure trough web.
But what i don’t understand, when i add extension trough freepbx, i got message “Call failed: Service Unavailable” (using x-lite). which i dont get it if i add extension trough sip.conf and extension.conf.

Simple, i can make a phone call (using X-Lite) if i add extension trough extension.conf but it won’t work if i add trough freepbx.

Any ide where i have to check. ??

2

Here are the message i got from asterisk console.

– Registered SIP ‘239’ at 10.1.8.239 port 52438 expires 3600
– Executing Macro(“SIP/239-09d73608”, “exten-vm|novm|239”) in new stack
– Executing Macro(“SIP/239-09d73608”, “user-callerid”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?report”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?start”) in new stack
– Executing Set(“SIP/239-09d73608”, “REALCALLERIDNUM=239”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “REALCALLERIDNUM is 239”) in new stack
– Executing Set(“SIP/239-09d73608”, “AMPUSER=239”) in new stack
– Executing Set(“SIP/239-09d73608”, “AMPUSERCIDNAME=239”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?report”) in new stack
– Executing Set(“SIP/239-09d73608”, “CALLERID(all)=239 <239>”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “Using CallerID “239” <239>”) in new stack
– Executing Set(“SIP/239-09d73608”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Set(“SIP/239-09d73608”, “VMBOX=novm”) in new stack
– Executing Set(“SIP/239-09d73608”, “EXTTOCALL=239”) in new stack
– Executing Set(“SIP/239-09d73608”, “CFUEXT=”) in new stack
– Executing Set(“SIP/239-09d73608”, “RT=”) in new stack
– Executing Macro(“SIP/239-09d73608”, “record-enable|239|IN”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0 > 0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing AGI(“SIP/239-09d73608”, “recordingcheck|20060913-120342|1158123822.11”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/239-09d73608”, “No recording needed”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “0?dolocaldial|1”) in new stack
– Executing Macro(“SIP/239-09d73608”, “dial||tr|239”) in new stack
– Executing AGI(“SIP/239-09d73608”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
– AGI Script dialparties.agi completed, returning 0
– Executing NoOp(“SIP/239-09d73608”, “Returned from dialparties with no extensions to call”) in new stack
– Executing NoOp(“SIP/239-09d73608”, "DIALSTATUS is ") in new stack
– Executing GosubIf(“SIP/239-09d73608”, “0?docfu|1”) in new stack
– Executing NoOp(“SIP/239-09d73608”, “Voicemail is novm”) in new stack
– Executing GotoIf(“SIP/239-09d73608”, “1?s-|1”) in new stack
– Goto (macro-exten-vm,s-,1)
– Executing Congestion(“SIP/239-09d73608”, “”) in new stack
== Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/239-09d73608’ in macro ‘exten-vm’
== Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/239-09d73608’

3

hy suse? Because the company decieded to use suse as linux distribution.
Which guide I have used? The one included in the freepbx package.
It could be a problem that the server is a P4 HT and there is a suse 10.1 x64 installed on it.
Someone told me once that asterisk doesn’t work in x64 distributions and I sall install the 32 bit one. It is true or not?

4

well, from the beggining I am using SuSE 10.1 x54 distribution,
the asterisk is the lattest: Asterisk 1.2.13
the freebpx is Version 2.1.3
I have installed all this from tabball after the INSTALL file from the freepbx tarball
And the only thing I set up is the 2 sip account for tests.

here is the log of the calling

Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Setting NAT on RTP to 0
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 214: Match Found
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Setting NAT on RTP to 0
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Checking SIP call limits for device 490
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: build_route: Contact hop: sip:[email protected]
Oct 26 11:52:07 DEBUG[5796] channel.c: Avoiding initial deadlock for 'SIP/490-b49033e0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Macro(“SIP/490-b49033e0”, “exten-vm|novm|491”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Macro(“SIP/490-b49033e0”, “user-callerid”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “0?report”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Not taking any branch
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “0?start”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Not taking any branch
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '490’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “REALCALLERIDNUM=490”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, “REALCALLERIDNUM is 490”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '490’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “AMPUSER=490”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'xxx’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “AMPUSERCIDNAME=xxx”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “0?report”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Not taking any branch
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “CALLERID(all)=xxx <490>”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '“xxx” <490>'
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, “Using CallerID “xxx” <490>”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “FROMCONTEXT=exten-vm”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “VMBOX=novm”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “EXTTOCALL=491”) in new stack
Oct 26 11:52:07 DEBUG[5900] db.c: Unable to find key ‘491’ in family 'CFU’
Oct 26 11:52:07 DEBUG[5900] func_db.c: DB: CFU/491 not found in database.
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is ''
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “CFUEXT=”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is '0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is ''
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Set(“SIP/490-b49033e0”, “RT=”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Macro(“SIP/490-b49033e0”, “record-enable|491|IN”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is ‘0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “0 > 0?2:4”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Goto (macro-record-enable,s,4)
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing AGI(“SIP/490-b49033e0”, “recordingcheck|20061026-115207|1161856327.1”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
Oct 26 11:52:07 VERBOSE[5900] logger.c: – AGI Script recordingcheck completed, returning 0
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, “No recording needed”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is ‘0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “0?dolocaldial|1”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Not taking any branch
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Macro(“SIP/490-b49033e0”, “dial||tr|491”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing AGI(“SIP/490-b49033e0”, “dialparties.agi”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
Oct 26 11:52:07 VERBOSE[5900] logger.c: – AGI Script dialparties.agi completed, returning 0
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, “Returned from dialparties with no extensions to call”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, "DIALSTATUS is ") in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is ‘0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is ‘0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is ‘0’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GosubIf(“SIP/490-b49033e0”, “0?docfu|1”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing NoOp(“SIP/490-b49033e0”, “Voicemail is novm”) in new stack
Oct 26 11:52:07 DEBUG[5900] pbx.c: Expression result is ‘1’
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing GotoIf(“SIP/490-b49033e0”, “1?s-|1”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Goto (macro-exten-vm,s-,1)
Oct 26 11:52:07 VERBOSE[5900] logger.c: – Executing Congestion(“SIP/490-b49033e0”, “”) in new stack
Oct 26 11:52:07 VERBOSE[5900] logger.c: == Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/490-b49033e0’ in macro ‘exten-vm’
Oct 26 11:52:07 VERBOSE[5900] logger.c: == Spawn extension (macro-exten-vm, s-, 1) exited non-zero on ‘SIP/490-b49033e0’
Oct 26 11:52:07 DEBUG[5900] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Oct 26 11:52:07 DEBUG[5900] 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 (‘2006-10-26 11:52:07’,’“xxx” <490>’,‘490’,‘491’,‘from-internal’, ‘SIP/490-b49033e0’,’’,‘Congestion’,’’,0,0,‘NO ANSWER’,3,’’,‘1161856327.1’)
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Failed to grab lock, trying again…
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '“xxx” <490>'
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '490’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '491’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'from-internal’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'SIP/490-b49033e0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '(null)'
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'Congestion’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '(null)'
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '2006-10-26 11:52:07’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '(null)'
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '2006-10-26 11:52:07’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '0’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'NO ANSWER’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is 'DOCUMENTATION’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '(null)'
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '1161856327.1’
Oct 26 11:52:07 DEBUG[5900] pbx.c: Function result is '(null)'
Oct 26 11:52:07 DEBUG[5900] chan_sip.c: update_call_counter(490) - decrement call limit counter
Oct 26 11:52:07 DEBUG[5807] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Response 215: Match Not Found

5

well I also have the same issue but noone can’t or want to give me an answer.
I don’t think that there is noone who managed to resolve this problem.

6

I do not see that you posted a problem

Freepbx works so if it is not working for you then we need to just find out where it is “broke”

So start off with some background info
What Version of asterisk / freepbx
Installed from AAH / TB ISO??
If not how?? What Version of Linux??

What are the errors??
LOG files??? you need to post them…

Have you hit the IRC and asked for help???

What have YOU DONE to help us help U???

7

I got the same problem as dungdung.
I’ve installed asterisk then freepbx as in the INSTALL file say.
I set up 3 sip extensions from freepbx andhen I call the one of the extenstions and it hang up without even calling the other extenstion as it asn’t registered or if hte extenstion has a vocemail set up I got automatically the vocemail of the ther extenstion. but it was.
in the cli I got the same output as dungdung.

8

Once more read my reply and post the info
If this is still on the SUSE install, that has a few gotta’s
And WHY suse…why not use centos which “works out of the box”

And unless you have ful-filled those added requirements needed to run freepbx on SuSe it is at all stop…

Go do this if you have not…
http://www.aussievoip.com/wiki/index.php?page=freePBX-SuSE

If you have you need to say so There are 100’s of Install guides so tell us which one you used

The more DETAILS you provide the better for us.
so do not use someone else thread info as YOURS…

Post your info…

unless the OP is running the extact same setup as you…then we could spend all day chasing our tails.

Put some effort into OK…

9

try iax softphone
http://www.laser.com/dante/diax/diax.html

10

It reads from the conf file…the database builds those files with the cmd above

11

Still not working… :frowning:

could you give me a clue where i should check.

-Running command as you suggested has no error
Checking for PEAR DB…OK
Checking for PEAR Console::Getopt…OK
Checking for /etc/amportal.conf…OK
Reading /etc/amportal.conf…OK
Reading /etc/asterisk/asterisk.conf…OK
Connecting to database…OK
Please Reload Asterisk by visiting http://10.1.1.26/admin

from CLI; if i look the database using “database show” it list all the extension i add trough the web.

i also could see it directly from mysql.

anything that i miss??

12

freepbx writes out the files

go to cli and run this
/var/lib/asterisk/bin/retrieve_conf
it is the cmd to pull the data from mysql…
did u check is see if mysql is running or no?..

13

mysql server is running. i could see the list of users from database.

/retrieve_conf should running from shell not from CLI isn’t it?

i suspect from this:
– AGI Script dialparties.agi completed, returning 0
– Executing NoOp(“SIP/239-09d73608”, “Returned from dialparties with no extensions to call”) in new stack
– Executing NoOp(“SIP/239-09d73608”, "DIALSTATUS is ") in new stack
– Executing GosubIf(“SIP/239-09d73608”, “0?docfu|1”) in new stack

but i don’t know how to trace it.
it’s like asterisk could read its own configuration from *.conf but it cann’t read from database on my case… :frowning: :frowning: