Hi,
I have just put in a tdm400p with 4 fxo modules and am trying to dial
out from x-lite to dial my mobile phone just to test.
The output in the asterisk console is like this
Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in
new stack
-- Goto (mobile,61400039953,1)
-- Executing Goto("SIP/2002-239b",
"localcall|61400039953|1") in
new stack
-- Goto (localcall,61400039953,1)
-- Executing Dial("SIP/2002-239b",
"ZAP/1/61400039953|60|r") in new
stack
-- Called 1/61400039953
-- Zap/1-1 answered SIP/2002-239b
-- Hungup 'Zap/1-1'
== Spawn extension (localcall, 61400039953, 1) exited non-zero on
'SIP/2002-239b'
It never tries to pick up the phone and dial out. I'm not sure if the
config is correct, but I can easily dial between x-lite clients, just
not get the pstn.
Can anyone see any glaring mistakes?
Any help is grealty appreciated.
Regards,
Greg
My extensions.conf part is this:
exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1)
[localcall] ; local calls by PSTN ?is a fixed charge, voip is per minute
exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r)
exten => _X.,2,Congestion
exten => _X.,3,Hangup
exten => _X.,103,Hangup
exten => _X.,104,Hangup
exten => _X.,105,Hangup
[mobile] ; Maybe be cheaper to route mobile calls differently to STD in
some cases
exten => _X.,1,Goto(localcall,${EXTEN},1)
zaptel.conf
fxsks=1-4
loadzone=au
defaultzone=au
channels=1-4
zapata.conf
[channels]
?
busydetect=1
busycount=7
?
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
?
usecallerid=yes
?
echocancel=yes
echocancelwhenbridged=yes
?
rxgain=0.0
txgain=0.0
?
group=1
pickupgroup=1-4
?
immediate=no
?
context=incomingcall
?
signalling=fxs_ks
callerid=asreceived
channel=1-4
On Fri, 18 Mar 2005 12:00:58 +1000, Greg <ghulands@framedphotographics.com> wrote:> Can anyone see any glaring mistakes?Yes.> My extensions.conf part is this: > > exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1)In Australia we don't prefix calls to mobiles with 61. You want something like: exten => _04XXXXXXXX,1,Goto(mobile,${EXTEN},1) If you're using a VoIP provider that requires 61, as well as routing calls through Zap where no 61 is required, you'll have to put in some logic to fix that up. -Shaun
I was just copy an example from somewhere. I made the change but the
mobile still doesn't ring. The line the card is attached to works fine.
here is the new output
Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in
new stack
-- Goto (mobile,0400039953,1)
-- Executing Goto("SIP/2002-4385",
"localcall|0400039953|1") in new
stack
-- Goto (localcall,0400039953,1)
-- Executing Dial("SIP/2002-4385",
"ZAP/1/0400039953|60|r") in new
stack
-- Called 1/0400039953
-- Zap/1-1 answered SIP/2002-4385
-- Hungup 'Zap/1-1'
== Spawn extension (localcall, 0400039953, 1) exited non-zero on
'SIP/2002-4385'
is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card
tries to make the call or when the card thinks it has established the
call?
Regards,
Greg
By the way, I'm on the Gold Coast.
On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote:
> Greg,
>
> Any reason why you are putting the country code on the front for a
> mobile
> call through pstn?
> (Unless you have something like an Ericsson F220M Fixed Cellular
> Terminal
> connected to it?)
>
> And you said the tdm400p never tries to pick up the phone?
> Have you connected a normal phone on the line and had a listen?
>
>
> Where is Aus are you? :o)
>
> Cheers
> Shane
>
>> -----Original Message-----
>> From: asterisk-users-bounces@lists.digium.com
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Greg
>> Sent: Friday, 18 March 2005 1:01 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [Asterisk-Users] Newbie can't dial out to pstn
>>
>> Hi,
>> I have just put in a tdm400p with 4 fxo modules and am trying
>> to dial out from x-lite to dial my mobile phone just to test.
>>
>> The output in the asterisk console is like this
>>
>> Executing Goto("SIP/2002-239b",
"mobile|61400039953|1") in new stack
>> -- Goto (mobile,61400039953,1)
>> -- Executing Goto("SIP/2002-239b",
>> "localcall|61400039953|1") in new stack
>> -- Goto (localcall,61400039953,1)
>> -- Executing Dial("SIP/2002-239b",
>> "ZAP/1/61400039953|60|r") in new stack
>> -- Called 1/61400039953
>> -- Zap/1-1 answered SIP/2002-239b
>> -- Hungup 'Zap/1-1'
>> == Spawn extension (localcall, 61400039953, 1) exited
>> non-zero on 'SIP/2002-239b'
>>
>> It never tries to pick up the phone and dial out. I'm not
>> sure if the config is correct, but I can easily dial between
>> x-lite clients, just not get the pstn.
>>
>> Can anyone see any glaring mistakes?
>>
>> Any help is grealty appreciated.
>>
>> Regards,
>> Greg
>>
>> My extensions.conf part is this:
>>
>> exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1)
>>
>> [localcall] ; local calls by PSTN ?is a fixed charge, voip is
>> per minute exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten =>
>> _X.,2,Congestion exten => _X.,3,Hangup exten =>
>> _X.,103,Hangup exten => _X.,104,Hangup exten => _X.,105,Hangup
>>
>> [mobile] ; Maybe be cheaper to route mobile calls differently
>> to STD in some cases exten => _X.,1,Goto(localcall,${EXTEN},1)
>>
>> zaptel.conf
>> fxsks=1-4
>> loadzone=au
>> defaultzone=au
>> channels=1-4
>>
>> zapata.conf
>> [channels]
>> ?
>> busydetect=1
>> busycount=7
>> ?
>> relaxdtmf=yes
>> callwaiting=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> cancallforward=yes
>> ?
>> usecallerid=yes
>> ?
>> echocancel=yes
>> echocancelwhenbridged=yes
>> ?
>> rxgain=0.0
>> txgain=0.0
>> ?
>> group=1
>> pickupgroup=1-4
>> ?
>> immediate=no
>> ?
>> context=incomingcall
>> ?
>> signalling=fxs_ks
>> callerid=asreceived
>> channel=1-4
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users@lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
What version of Asterisk? If this is not Asterisk@home you may want to install it and start over. It eases many of the problems experienced by newbs when learning *. Otherwise, make sure you use the ztcfg -vvvv so you can see some error verbosity. You may need to recompile your zaptel stuff. Just make sure you follow the instructions and recompile asterisk after. Regards, Wliey -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Greg Sent: Thursday, March 17, 2005 10:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Shane Dalgleish Subject: Re: [Asterisk-Users] Newbie can't dial out to pstn I have just run ztcfg and got these errors: # ztcfg Notice: Configuration file is /etc/zaptel.conf line 209: Cannot get number of tones chanel 1 line 209: Cannot init tones chanel 1 line 209: Cannot get number of tones chanel 2 line 209: Cannot init tones chanel 2 line 209: Cannot get number of tones chanel 3 line 209: Cannot init tones chanel 3 line 209: Cannot get number of tones chanel 4 line 209: Cannot init tones chanel 4 What would these mean. I searched the archives and couldn't find these errors. Greg On 18/03/2005, at 1:24 PM, Greg wrote:> I was just copy an example from somewhere. I made the change but the > mobile still doesn't ring. The line the card is attached to works > fine. here is the new output > > Executing Goto("SIP/2002-4385", "mobile|0400039953|1") in new stack > -- Goto (mobile,0400039953,1) > -- Executing Goto("SIP/2002-4385", "localcall|0400039953|1") in > new stack > -- Goto (localcall,0400039953,1) > -- Executing Dial("SIP/2002-4385", "ZAP/1/0400039953|60|r") in new > stack > -- Called 1/0400039953 > -- Zap/1-1 answered SIP/2002-4385 > -- Hungup 'Zap/1-1' > == Spawn extension (localcall, 0400039953, 1) exited non-zero on > 'SIP/2002-4385' > > is this line -- Zap/1-1 answered SIP/2002-4385 displayed when the card > tries to make the call or when the card thinks it has established the > call? > > Regards, > Greg > > By the way, I'm on the Gold Coast. > > On 18/03/2005, at 12:32 PM, Shane Dalgleish wrote: > >> Greg, >> >> Any reason why you are putting the country code on the front for a >> mobile call through pstn? >> (Unless you have something like an Ericsson F220M Fixed Cellular >> Terminal connected to it?) >> >> And you said the tdm400p never tries to pick up the phone? >> Have you connected a normal phone on the line and had a listen? >> >> >> Where is Aus are you? :o) >> >> Cheers >> Shane >> >>> -----Original Message----- >>> From: asterisk-users-bounces@lists.digium.com >>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Greg >>> Sent: Friday, 18 March 2005 1:01 PM >>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>> Subject: [Asterisk-Users] Newbie can't dial out to pstn >>> >>> Hi, >>> I have just put in a tdm400p with 4 fxo modules and am trying to >>> dial out from x-lite to dial my mobile phone just to test. >>> >>> The output in the asterisk console is like this >>> >>> Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack >>> -- Goto (mobile,61400039953,1) >>> -- Executing Goto("SIP/2002-239b", >>> "localcall|61400039953|1") in new stack >>> -- Goto (localcall,61400039953,1) >>> -- Executing Dial("SIP/2002-239b", >>> "ZAP/1/61400039953|60|r") in new stack >>> -- Called 1/61400039953 >>> -- Zap/1-1 answered SIP/2002-239b >>> -- Hungup 'Zap/1-1' >>> == Spawn extension (localcall, 61400039953, 1) exited non-zero on >>> 'SIP/2002-239b' >>> >>> It never tries to pick up the phone and dial out. I'm not sure if >>> the config is correct, but I can easily dial between x-lite clients, >>> just not get the pstn. >>> >>> Can anyone see any glaring mistakes? >>> >>> Any help is grealty appreciated. >>> >>> Regards, >>> Greg >>> >>> My extensions.conf part is this: >>> >>> exten => _04XXXXXXXX,1,GoTo(mobile,61${EXTEN:1},1) >>> >>> [localcall] ; local calls by PSTN ?is a fixed charge, voip is per >>> minute exten => _X.,1,Dial(ZAP/1/${EXTEN},60,r) exten => >>> _X.,2,Congestion exten => _X.,3,Hangup exten => _X.,103,Hangup exten >>> => _X.,104,Hangup exten => _X.,105,Hangup >>> >>> [mobile] ; Maybe be cheaper to route mobile calls differently to STD >>> in some cases exten => _X.,1,Goto(localcall,${EXTEN},1) >>> >>> zaptel.conf >>> fxsks=1-4 >>> loadzone=au >>> defaultzone=au >>> channels=1-4 >>> >>> zapata.conf >>> [channels] >>> ? >>> busydetect=1 >>> busycount=7 >>> ? >>> relaxdtmf=yes >>> callwaiting=yes >>> callwaitingcallerid=yes >>> threewaycalling=yes >>> transfer=yes >>> cancallforward=yes >>> ? >>> usecallerid=yes >>> ? >>> echocancel=yes >>> echocancelwhenbridged=yes >>> ? >>> rxgain=0.0 >>> txgain=0.0 >>> ? >>> group=1 >>> pickupgroup=1-4 >>> ? >>> immediate=no >>> ? >>> context=incomingcall >>> ? >>> signalling=fxs_ks >>> callerid=asreceived >>> channel=1-4 >>> >>> _______________________________________________ >>> Asterisk-Users mailing list >>> Asterisk-Users@lists.digium.com >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users