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uk.telecom.voip (UK VOIP) (uk.telecom.voip) Discussion of topics relevant to packet based voice technologies including Voice over IP (VoIP), Fax over IP (FoIP), Voice over Frame Relay (VoFR), Voice over Broadband (VoB) and Voice on the Net (VoN) as well as service providers, hardware and software for use with these technologies. Advertising is not allowed.

[email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))



 
 
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  #1  
Old November 14th 07, 02:54 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

Hello all i have set up my [email protected] 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

  #2  
Old November 14th 07, 03:34 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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 wrote:
Hello all i have set up my [email protected] 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



  #3  
Old November 14th 07, 04:19 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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

  #4  
Old November 15th 07, 11:29 AM posted to uk.telecom.voip
Desk Rabbit
external usenet poster
 
Posts: 169
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME2)))

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?
  #5  
Old November 15th 07, 04:02 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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?

  #6  
Old November 15th 07, 08:39 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

phpguy explained on 15/11/2007 :

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?


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


Post the contents of your sip.conf here.......at the bottom of your
reply, though!


  #7  
Old November 15th 07, 09:20 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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
  #8  
Old November 15th 07, 09:38 PM posted to uk.telecom.voip
Jono
external usenet poster
 
Posts: 1,539
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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


  #9  
Old November 15th 07, 10:06 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

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

  #10  
Old November 15th 07, 10:11 PM posted to uk.telecom.voip
phpguy
external usenet poster
 
Posts: 32
Default [email protected] TRUNKS AND OUTGOING RULES PROBLEM !! (((VOLUME 2)))

================================================== =======================
================================================== =======================
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'
 




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