ASTERISK@HOME TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

Hello all i have set up my astyerisk@home system using also a zaptel
telephony card for my landline, when i call in i can hear my music on
hold fine and extensions ring but i cannot dialout either using ZAP
(my landline) or SIP

the settings are

i created 2 trunks with sip and ZAP and 2 outbound routings with dial
patterns 99|. for zap (landline) and 88|. to select the sip provider.
Both return busy when i dial out like 99+a number or 88+a number

why does this happen ?

cheers

sorry cancel the below only the SIP TRUNK doesnt work and gives a busy
tone

all settigns are correct though such as ip login and paossword

i dont understand :/

On 14 Nov, 16:54, phpguy <ksma...@hotmail.com> wrote:
> Hello all i have set up my astyerisk@home system using also a zaptel
> telephony card for my landline, when i call in i can hear my music on
> hold fine and extensions ring but i cannot dialout either using ZAP
> (my landline) or SIP
>
> the settings are
>
> i created 2 trunks with sip and ZAP and 2 outbound routings with dial
> patterns 99|. for zap (landline) and 88|. to select the sip provider.
> Both return busy when i dial out like 99+a number or 88+a number
>
> why does this happen ?
>
> cheers

another add-on, when i try dialing with the same provider (voipbuster)
from my sip phone directly it works but it wont work via asterisk it
gives busy signal and logs show channel unavailable for some reason ?

cheers

phpguy wrote:
> another add-on, when i try dialing with the same provider (voipbuster)
> from my sip phone directly it works but it wont work via asterisk it
> gives busy signal and logs show channel unavailable for some reason ?
>
> cheers
>
Wild guess, is your Asterisk box behind a NAT router?

yes but so are all ip phones that do work when i enter the
voipbuster.com settings to them manually inside their own menu

so what do i do now

cheers

Desk Rabbit wrote:

> phpguy wrote:
> > another add-on, when i try dialing with the same provider (voipbuster)
> > from my sip phone directly it works but it wont work via asterisk it
> > gives busy signal and logs show channel unavailable for some reason ?
> >
> > cheers
> >
> Wild guess, is your Asterisk box behind a NAT router?

ok here it is

; Note: If your SIP devices are behind a NAT and your Asterisk
; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

[general]

port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf

no luck mate

a female voice says ALL CIRCUITS ARE BUSY NOW TRY AGAIN LATER

SO STRANGE

Jono wrote:

> phpguy formulated on Thursday :
> > ok here it is
> >
> > ; Note: If your SIP devices are behind a NAT and your Asterisk
> > ; server isn't, try adding "nat=1" to each peer definition to
> > ; solve translation problems.
> >
> > [general]
> >
> > port = 5060 ; Port to bind to (SIP is 5060)
> > bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > context = from-sip-external ; Send unknown SIP callers to this context
> > callerid = Unknown
> >
> > #include sip_nat.conf
> > #include sip_custom.conf
> > #include sip_additional.conf
>
> [general]
>
> port = 5060 ; Port to bind to (SIP is 5060)
> bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
> disallow=all
> allow=ulaw
> allow=alaw
> allow=gsm
> ;context = from-sip-external ; Send unknown SIP callers to this context
> context = from-trunk
> ;defaultexpirey = 600 ; include this only if necessary
> ;maxexpirey = 3600 ; include this only if necessary
> progressinband = yes
> ;dtmfmode=auto
>
> callerid = Unknown
> externip = myPUBLICipADDRESS/DynDNShostname
> localnet=192.168.1.0/255.255.255.0
> nat=yes
>
> #include sip_nat.conf
> #include sip_custom.conf
> #include sip_additional.conf

================================================== =======================
================================================== =======================
asterisk1*CLI> Verbosity is at least 3
Connected to Asterisk 1.0.9 currently running on asterisk1 (pid =
1313)
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "1?3:2)") in
new stack
-- Goto (macro-dialout-trunk,s,3)
Verbosity is at least 3
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0 > 0?2:4") in
new stack
-- Goto (macro-record-enable,s,4)
-- Executing Macro("SIP/200-2e40", "dialout-trunk|2|
0030210XXXXXXXX|") in new stack
asterisk1*CLI> -- Goto (macro-record-enable,s,5)
-- Executing GotoIf("SIP/200-2e40", "13:2)") in new stack
asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
key=200
-- Goto (macro-dialout-trunk,s,3)
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
"CALLFILENAME=OUT200-19990107-023107-915694267.2") in new stack
-- Executing Goto("SIP/200-2e40", "s|14") in new stack
-- Executing Macro("SIP/200-2e40", "record-enable|200|OUT") in new
stack
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0?15:99") in
new stack
-- Executing GotoIf("SIP/200-2e40", "0 > 02:4") in new stack
asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "NO RECORDING
NEEDED") in new stack
-- Executing GotoIf("SIP/200-2e40", "1?7") in new stack
-- Goto (macro-record-enable,s,4)
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "0?9") in new
stack
-- Executing GotoIf("SIP/200-2e40", "15:8") in new stack
asterisk1*CLI> -- Executing SetGroup("SIP/200-2e40", "OUT_2") in
new stack
-- Goto (macro-record-enable,s,5)
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
"DIAL_NUMBER=XXXXXXXX") in new stack
-- Executing DBget("SIP/200-2e40", "RecEnable=RECORD-OUT/200") in
new stack
asterisk1*CLI> -- DBget: varname=RecEnable, family=RECORD-OUT,
key=200
asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- DBget: Value not found in database.
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
"OUTNUM=XXXXXXXXXXX") in new stack
-- Executing Cut("SIP/200-2e40", "custom=OUT_2|:|1") in new stack
-- Executing SetVar("SIP/200-2e40",
"CALLFILENAME=OUT200-19990107-023107-915694267.2") in new stack
asterisk1*CLI> -- Executing Goto("SIP/200-2e40", "s|14") in new
stack
asterisk1*CLI> -- Goto (macro-record-enable,s,14)
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "015:99") in
new stack
asterisk1*CLI> -- Goto (macro-record-enable,s,99)
asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "NO RECORDING
NEEDED") in new stack
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "17") in new
stack
asterisk1*CLI> -- Goto (macro-dialout-trunk,s,7)
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "09") in new
stack
asterisk1*CLI> -- Executing SetCallerID("SIP/200-2e40",
"asiawatcher") in new stack
asterisk1*CLI> -- Executing SetGroup("SIP/200-2e40", "OUT_2") in
new stack
asterisk1*CLI> -- Executing CheckGroup("SIP/200-2e40", "") in new
stack
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
"DIAL_NUMBER=XXXXXXXXXXXXX") in new stack
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40", "DIAL_TRUNK=2")
in new stack
asterisk1*CLI> -- Executing AGI("SIP/200-2e40", "fixlocalprefix")
in new stack
asterisk1*CLI> -- Launched AGI Script /var/lib/asterisk/agi-bin/
fixlocalprefix
asterisk1*CLI> fixlocalprefix: Could not parse /etc/asterisk/
localprefixes.conf
asterisk1*CLI> -- AGI Script fixlocalprefix completed, returning 0
asterisk1*CLI> -- Executing SetVar("SIP/200-2e40",
"OUTNUM=XXXXXXXXXXX") in new stack
asterisk1*CLI> -- Executing Cut("SIP/200-2e40", "custom=OUT_2|:|
1") in new stack
asterisk1*CLI> -- Executing GotoIf("SIP/200-2e40", "019") in new
stack
asterisk1*CLI> -- Executing Dial("SIP/200-2e40", "SIP/
voipbusterfinal/XXXXXXXXXXXXXX") in new stack
asterisk1*CLI> == Everyone is busy/congested at this time
asterisk1*CLI> -- Executing Goto("SIP/200-2e40", "s-CHANUNAVAIL|
1") in new stack
asterisk1*CLI> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
asterisk1*CLI> -- Executing NoOp("SIP/200-2e40", "Dial failed due
to CHANUNAVAIL") in new stack
asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "outisbusy") in
new stack
asterisk1*CLI> -- Executing Playback("SIP/200-2e40", "allison7/all-
circuits-busy-now") in new stack
asterisk1*CLI> -- Playing 'allison7/all-circuits-busy-
now' (language 'en')
asterisk1*CLI> -- Executing Playback("SIP/200-2e40", "allison7/pls-
try-call-later") in new stack
asterisk1*CLI> -- Playing 'allison7/pls-try-call-later' (language
'en')
asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "hangupcall") in
new stack
asterisk1*CLI> -- Executing ResetCDR("SIP/200-2e40", "w") in new
stack
asterisk1*CLI> -- Executing NoCDR("SIP/200-2e40", "") in new stack
asterisk1*CLI> -- Executing Wait("SIP/200-2e40", "5") in new stack
asterisk1*CLI> -- Executing Hangup("SIP/200-2e40", "") in new
stack
asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 4) exited
non-zero on 'SIP/200-2e40' in macro 'hangupcall'
asterisk1*CLI> == Spawn extension (macro-outisbusy, s, 3) exited non-
zero on 'SIP/200-2e40' in macro 'outisbusy'
asterisk1*CLI> == Spawn extension (from-internal, 8800302107292105,
2) exited non-zero on 'SIP/200-2e40'
asterisk1*CLI> -- Executing Macro("SIP/200-2e40", "hangupcall") in
new stack
asterisk1*CLI> -- Executing ResetCDR("SIP/200-2e40", "w") in new
stack
asterisk1*CLI> -- Executing NoCDR("SIP/200-2e40", "") in new stack
asterisk1*CLI> -- Executing Wait("SIP/200-2e40", "5") in new stack
asterisk1*CLI> == Spawn extension (macro-hangupcall, s, 3) exited
non-zero on 'SIP/200-2e40' in macro 'hangupcall'
asterisk1*CLI> == Spawn extension (from-internal, h, 1) exited non-
zero on 'SIP/200-2e40'

yes i restarted asterisk nothing

trunk details are the default SIP ones only i replace my username/
password and sip domain

cheers