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ASTERISK@HOME TRUNKS AND OUTGOING RULES PROBLEM !!

 
 
phpguy
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Posts: n/a
 
      11-10-2007
hello all i got a simple question im using asterisk@home and i got x2
trunks one zap trunk that is using my normal landline for inbound and
outgoing calls and i also got a sip trunk with voipbuster

im trying to set when i dial 99+a number to use the zap trunk and when
i dial 88+a number to dial the specified number over the SIP trunk

in few words

99 077123456 should dial a uk mobile over the zap trunk connected to
my landline using a fxo card and when i dial 88 003031072XXXX should
dial greece over the SIP trunk

im trying to do that with the rules both in trunk and outgoing routing
of:

99|. for zap and 88|. for sip

problem is it works for zap but for sip it wont work

what am i doing wrong please ?

cheers

 
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alexd
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      11-10-2007
phpguy wrote:

> im trying to do that with the rules both in trunk and outgoing routing
> of:
>
> 99|. for zap and 88|. for sip
>
> problem is it works for zap but for sip it wont work
>
> what am i doing wrong please ?


What happens when it doesn't work? My first suggestion would be to look at
the Asterisk console ['asterisk -rc'] and see what it says. If it doesn't
say much, then run 'set verbose 5', and try again. Also, does your SIP
trunk work at all, ie without fancy routing? Can you receive calls on it
[if applicable]?

--
<http://ale.cx/> (AIM:troffasky) ((E-Mail Removed))
18:34:38 up 8 days, 11:19, 3 users, load average: 0.16, 0.17, 0.17
50,000 watts of funking power

 
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phpguy
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      11-10-2007

================================================== =======================
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
1313)
Verbosity is at least 3
asterisk1*CLI> set verbose 5
Verbosity was 3 and is now 5
-- Executing Macro("SIP/201-f9d3", "dialout-trunk|3|
00302107292105|") in new stack
-- Executing GotoIf("SIP/201-f9d3", "1?3:2)") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/201-f9d3", "record-enable|201|OUT") in new
stack
-- Executing GotoIf("SIP/201-f9d3", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/201-f9d3", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/201-f9d3", "RecEnable=RECORD-OUT/201") in
new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=201
-- DBget: set variable RecEnable to DISABLED
-- Executing SetVar("SIP/201-f9d3",
"CALLFILENAME=OUT201-19990101-232229-915
250949.2") in new stack
-- Executing Goto("SIP/201-f9d3", "s|14") in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf("SIP/201-f9d3", "0?15:99") in new stack
-- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
stack
-- Executing GotoIf("SIP/201-f9d3", "1?7") in new stack
-- Goto (macro-dialout-trunk,s,7)
-- Executing GotoIf("SIP/201-f9d3", "1?9") in new stack
-- Goto (macro-dialout-trunk,s,9)
-- Executing SetGroup("SIP/201-f9d3", "OUT_3") in new stack
-- Executing CheckGroup("SIP/201-f9d3", "") in new stack
-- Executing SetVar("SIP/201-f9d3", "DIAL_NUMBER=00302107292105")
in new sta ck
-- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3") in new stack
-- Executing AGI("SIP/201-f9d3", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
asterisk1*CLI>
! abort add agi cdr database
debug dont dump exit extensions help
iax2 include init load local logger
meetme mgcp no pri quit reload
remove restart set show sip skinny
soft stop unload zap
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar("SIP/201-f9d3", "OUTNUM=00302107292105") in
new stack
-- Executing Cut("SIP/201-f9d3", "custom=OUT_3|1") in new stack
-- Executing GotoIf("SIP/201-f9d3", "0?19") in new stack
-- Executing Dial("SIP/201-f9d3", "SIP/
voipbuster2/00302107292105") in new s tack
== Everyone is busy/congested at this time
-- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing NoOp("SIP/201-f9d3", "Dial failed due to
CHANUNAVAIL") in new s tack
-- Executing Macro("SIP/201-f9d3", "outisbusy") in new stack
-- Executing Playback("SIP/201-f9d3", "allison7/all-circuits-busy-
now") in n ew stack
-- Playing 'allison7/all-circuits-busy-now' (language 'en')
-- Executing Playback("SIP/201-f9d3", "allison7/pls-try-call-
later") in new stack
-- Playing 'allison7/pls-try-call-later' (language 'en')
-- Executing Macro("SIP/201-f9d3", "hangupcall") in new stack
-- Executing ResetCDR("SIP/201-f9d3", "w") in new stack
-- Executing NoCDR("SIP/201-f9d3", "") in new stack
-- Executing Wait("SIP/201-f9d3", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
201-f9d3' in macro 'hangupcall'
== Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'SIP/
201-f9d3' i n macro 'outisbusy'
== Spawn extension (from-internal, 800302107292105, 2) exited non-
zero on 'SIP /201-f9d3'
-- Executing Macro("SIP/201-f9d3", "hangupcall") in new stack
-- Executing ResetCDR("SIP/201-f9d3", "w") in new stack
-- Executing NoCDR("SIP/201-f9d3", "") in new stack
-- Executing Wait("SIP/201-f9d3", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/
201-f9d3' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/201-
f9d3'
asterisk1*CLI>
asterisk1*CLI> Connected to Asterisk 1.0.9 currently running on
asterisk1 (pid = 1313)
Verbosity is at least 3
asterisk1*CLI> set verbose 5
Verbosity was 3 and is now 5
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
1313)
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "1?3:2)") in
new stack
Verbosity is at least 3
asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "record-enable|
201|OUT") in new stack
asterisk1*CLI> set verbose 5
asterisk1*CLI> -- Goto (macro-record-enable,s,4)
-- Executing GotoIf("SIP/201-f9d3", "1?5:8") in new stack
-- Goto (macro-record-enable,s,5)
-- Executing DBget("SIP/201-f9d3", "RecEnable=RECORD-OUT/201") in
new stack
-- DBget: varname=RecEnable, family=RECORD-OUT, key=201
-- DBget: set variable RecEnable to DISABLED
-- Executing SetVar("SIP/201-f9d3",
"CALLFILENAME=OUT201-19990101-232229-915
250949.2") in new stack
-- Executing Goto("SIP/201-f9d3", "s|14") in new stack
Verbosity was 3 and is now 5
asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "dialout-trunk|3|
00302107292105|") in new stack
asterisk1*CLI> -- Goto (macro-record-enable,s,99)
-- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
stack
-- Executing GotoIf("SIP/201-f9d3", "1?7") in new stack
-- Executing GotoIf("SIP/201-f9d3", "13:2)") in new stack
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "1?9") in new
stack
-- Goto (macro-dialout-trunk,s,3)
asterisk1*CLI> -- Executing SetGroup("SIP/201-f9d3", "OUT_3") in
new stack
-- Executing CheckGroup("SIP/201-f9d3", "") in new stack
-- Executing Macro("SIP/201-f9d3", "record-enable|201|OUT") in new
stack
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "0 > 02:4") in
new stack
asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3")
in new stack
-- Goto (macro-record-enable,s,4)
asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
fixlocalprefix
asterisk1*CLI>
! abort add agi cdr database
debug dont dump exit extensions help
-- Executing GotoIf("SIP/201-f9d3", "15:8") in new stack
asterisk1*CLI> meetme mgcp no pri
quit reload
remove restart set show sip skinny
-- Goto (macro-record-enable,s,5)
asterisk1*CLI> -- Executing DBget("SIP/201-f9d3",
"RecEnable=RECORD-OUT/201") in new stack
asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
key=201
asterisk1*CLI> -- DBget: set variable RecEnable to DISABLED
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "0?19") in new
stack
-- Executing SetVar("SIP/201-f9d3",
"CALLFILENAME=OUT201-19990101-232229-915
250949.2") in new stack
asterisk1*CLI> -- Executing Dial("SIP/201-f9d3", "SIP/
voipbuster2/00302107292105") in new s tack
== Everyone is busy/congested at this time
-- Executing Goto("SIP/201-f9d3", "s|14") in new stack
asterisk1*CLI> -- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|
1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Goto (macro-record-enable,s,14)
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "015:99") in
new stack
asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
now' (language 'en')
-- Goto (macro-record-enable,s,99)
asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/pls-
try-call-later") in new stack
-- Executing NoOp("SIP/201-f9d3", "NO RECORDING NEEDED") in new
stack
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "17") in new
stack
asterisk1*CLI> -- Goto (macro-dialout-trunk,s,7)
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "19") in new
stack
asterisk1*CLI> -- Goto (macro-dialout-trunk,s,9)
asterisk1*CLI> -- Executing SetGroup("SIP/201-f9d3", "OUT_3") in
new stack
asterisk1*CLI> -- Executing CheckGroup("SIP/201-f9d3", "") in new
stack
asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3",
"DIAL_NUMBER=00302107292105") in new sta ck
asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3", "DIAL_TRUNK=3")
in new stack
asterisk1*CLI> -- Executing AGI("SIP/201-f9d3", "fixlocalprefix")
in new stack
asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
fixlocalprefix
asterisk1*CLI> asterisk1*CLI>
asterisk1*CLI> ! abort add agi
cdr database
/bin/sh: line 1: abort: command not found
asterisk1*CLI> debug dont dump exit
extensions help
asterisk1*CLI> iax2 include init load
local logger
asterisk1*CLI> meetme mgcp no pri
quit reload
asterisk1*CLI> remove restart set show
sip skinny
asterisk1*CLI> soft stop unload zap
asterisk1*CLI> -- AGI Script fixlocalprefix completed, returning 0
asterisk1*CLI> -- Executing SetVar("SIP/201-f9d3",
"OUTNUM=00302107292105") in new stack
asterisk1*CLI> -- Executing Cut("SIP/201-f9d3", "custom=OUT_3|
1") in new stack
asterisk1*CLI> -- Executing GotoIf("SIP/201-f9d3", "019") in new
stack
asterisk1*CLI> -- Executing Dial("SIP/201-f9d3", "SIP/
voipbuster2/00302107292105") in new s tack
asterisk1*CLI> == Everyone is busy/congested at this time
asterisk1*CLI> -- Executing Goto("SIP/201-f9d3", "s-CHANUNAVAIL|
1") in new stack
asterisk1*CLI> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
asterisk1*CLI> -- Executing NoOp("SIP/201-f9d3", "Dial failed due
to CHANUNAVAIL") in new s tack
asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "outisbusy") in
new stack
asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/all-
circuits-busy-now") in n ew stack
asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
now' (language 'en')
asterisk1*CLI> -- Executing Playback("SIP/201-f9d3", "allison7/pls-
try-call-later") in new stack
asterisk1*CLI> -- Playing 'allison7/pls-try-call-later' (language
'en')
asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "hangupcall") in
new stack
asterisk1*CLI> -- Executing ResetCDR("SIP/201-f9d3", "w") in new
stack
asterisk1*CLI> -- Executing NoCDR("SIP/201-f9d3", "") in new stack
asterisk1*CLI> -- Executing Wait("SIP/201-f9d3", "5") in new stack
asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/201-f9d3' in macro
'hangupcall'
asterisk1*CLI> == Spawn extension (macro-outisbusy, s, 3) exited non-
zero on 'SIP/201-f9d3' i n macro 'outisbusy'
asterisk1*CLI> == Spawn extension (from-internal, 800302107292105,
2) exited non-zero on 'SIP /201-f9d3'
asterisk1*CLI> -- Executing Macro("SIP/201-f9d3", "hangupcall") in
new stack
asterisk1*CLI> -- Executing ResetCDR("SIP/201-f9d3", "w") in new
stack
asterisk1*CLI> -- Executing NoCDR("SIP/201-f9d3", "") in new stack
asterisk1*CLI> -- Executing Wait("SIP/201-f9d3", "5") in new stack
asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/201-f9d3' in macro
'hangupcall'
asterisk1*CLI> == Spawn extension (from-internal, h, 1) exited non-
zero on 'SIP/201-f9d3'

 
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phpguy
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      11-10-2007
thanks for your repl;y i pasted what you requested

i get a busy tone or sometimes a timeout


 
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Paul Hayes
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      11-12-2007
phpguy wrote:
[snip]

voipbuster2/00302107292105") in new s tack
== Everyone is busy/congested at this time

Your service provider is returning busy when you send that number to
them. Nothing wrong with your dialplan by the looks of it but you
should check your sip.conf settings for your Voipbuster account and that
your account actually works if you just use the same settings directly
on an IP phone.

cheers,
Paul.

--
Working Email:

paul-at-polog40-dot-co-dot-uk
 
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