Henrik Woffinden
2006-Aug-27 08:50 UTC
[asterisk-users] Cannot dial out through SIP provider
Hi,
I'm running Asterisk 1.2.10 bristuffed.
Asterisk is registring perfectly against my provider (musimi.dk), and
incoming calls comes in and are routed fine to either internal ZAP
(ISDN BRI) and/or SIP.
But....
I can't dial out via SIP (musimi)
sip.conf:
[musimi]
type=friend
host=musimi.dk
username=xxxxxxxx
fromuser=xxxxxxxx
secret=xxxxxxxxxx
domain=musimi.dk
fromdomain=musimi.dk
context=from-sip
;nat=yes
;canreinvite=no
insecure=very
dtmfmode=rfc2833
[9999]
type=friend
context=internal
username=9999
secret=xxxxxxxx
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
callerid="Henrik Woffinden" <9999>
nat=yes
qualify=yes
insecure=very
;mailbox=9999@from-sip
extensions.conf:
[internal]
;exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN},,)
exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi,,)
exten => _XXXXXXXX,n,Hangup
If I want to dial out via ISDN (Zap which is commented out above), then
it works ok, but via SIP I get the following error message (my own
number is xxxxxxxx and the number I dial is yyyyyyyy - which is a normal
mobile):
-- Registered SIP '9999' at 192.168.9.9 port 29796 expires 3600
-- Executing Dial("SIP/9999-09f2eb28",
"SIP/yyyyyyyy@musimi||") in new stack
-- Called yyyyyyyy@musimi
Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
Failed to authenticate on INVITE to '"Henrik Woffinden"
<sip:xxxxxxxx@musimi.dk>;tag=as06ed5480'
-- SIP/musimi-09f34188 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("SIP/9999-09f2eb28", "") in new stack
== Spawn extension (internal, yyyyyyyy, 2) exited non-zero on
'SIP/9999-09f2eb28'
I hope somebody can tell me what I'm doing wrong here.
--
Med venlig hilsen / Best regards,
Henrik Woffinden
Shouldn't the line:
exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi,,)
be:
exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi.dk,,)
note the .dk in the second one...
Also I don't see a register line in your sip.conf. In the [general]
section I would have expected something like:
register=><a number>:<a password>@musimi.dk/musimi
I mention this because I had a problem where the domain name didn't
resolve, so I had to change the register line to use a dotted IP
address like this:
register=><a number>:<a password>@999.999.999.999/musimi
Don't laugh, it's the only way I could get it to work!!
Yours,
H
On 8/27/06, Henrik Woffinden <hw@nitramlexa.com>
wrote:> Hi,
>
> I'm running Asterisk 1.2.10 bristuffed.
> Asterisk is registring perfectly against my provider (musimi.dk), and
> incoming calls comes in and are routed fine to either internal ZAP
> (ISDN BRI) and/or SIP.
> But....
> I can't dial out via SIP (musimi)
>
> sip.conf:
> [musimi]
> type=friend
> host=musimi.dk
> username=xxxxxxxx
> fromuser=xxxxxxxx
> secret=xxxxxxxxxx
> domain=musimi.dk
> fromdomain=musimi.dk
> context=from-sip
> ;nat=yes
> ;canreinvite=no
> insecure=very
> dtmfmode=rfc2833
>
> [9999]
> type=friend
> context=internal
> username=9999
> secret=xxxxxxxx
> host=dynamic
> canreinvite=no
> dtfmode=rfc2833
> disallow=all
> allow=ulaw
> callerid="Henrik Woffinden" <9999>
> nat=yes
> qualify=yes
> insecure=very
> ;mailbox=9999@from-sip
>
> extensions.conf:
> [internal]
> ;exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN},,)
> exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi,,)
> exten => _XXXXXXXX,n,Hangup
>
>
> If I want to dial out via ISDN (Zap which is commented out above), then
> it works ok, but via SIP I get the following error message (my own
> number is xxxxxxxx and the number I dial is yyyyyyyy - which is a normal
> mobile):
>
> -- Registered SIP '9999' at 192.168.9.9 port 29796 expires 3600
> -- Executing Dial("SIP/9999-09f2eb28",
"SIP/yyyyyyyy@musimi||") in new stack
> -- Called yyyyyyyy@musimi
> Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite:
> Failed to authenticate on INVITE to '"Henrik Woffinden"
> <sip:xxxxxxxx@musimi.dk>;tag=as06ed5480'
> -- SIP/musimi-09f34188 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing Hangup("SIP/9999-09f2eb28", "") in new
stack
> == Spawn extension (internal, yyyyyyyy, 2) exited non-zero on
> 'SIP/9999-09f2eb28'
>
>
> I hope somebody can tell me what I'm doing wrong here.
>
> --
> Med venlig hilsen / Best regards,
>
> Henrik Woffinden
>
>
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----- Original Message ----- From: "Henrik Woffinden" <hw@nitramlexa.com> To: <asterisk-users@lists.digium.com> Sent: Sunday, August 27, 2006 11:50 AM Subject: [asterisk-users] Cannot dial out through SIP provider> Hi, > > I'm running Asterisk 1.2.10 bristuffed. > Asterisk is registring perfectly against my provider (musimi.dk), and > incoming calls comes in and are routed fine to either internal ZAP > (ISDN BRI) and/or SIP. > But.... > I can't dial out via SIP (musimi) > > sip.conf: > [musimi] > type=friend > host=musimi.dk > username=xxxxxxxx > fromuser=xxxxxxxx > secret=xxxxxxxxxx > domain=musimi.dk > fromdomain=musimi.dk > context=from-sip > ;nat=yes > ;canreinvite=no > insecure=very > dtmfmode=rfc2833 > > [9999] > type=friend > context=internal > username=9999 > secret=xxxxxxxx > host=dynamic > canreinvite=no > dtfmode=rfc2833 > disallow=all > allow=ulaw > callerid="Henrik Woffinden" <9999> > nat=yes > qualify=yes > insecure=very > ;mailbox=9999@from-sip > > extensions.conf: > [internal] > ;exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN},,) > exten => _XXXXXXXX,1,Dial(SIP/${EXTEN}@musimi,,) > exten => _XXXXXXXX,n,Hangup > > > If I want to dial out via ISDN (Zap which is commented out above), then > it works ok, but via SIP I get the following error message (my own > number is xxxxxxxx and the number I dial is yyyyyyyy - which is a normal > mobile): > > -- Registered SIP '9999' at 192.168.9.9 port 29796 expires 3600 > -- Executing Dial("SIP/9999-09f2eb28", "SIP/yyyyyyyy@musimi||") in new > stack > -- Called yyyyyyyy@musimi > Aug 27 16:38:41 NOTICE[23387]: chan_sip.c:9845 handle_response_invite: > Failed to authenticate on INVITE to '"Henrik Woffinden" > <sip:xxxxxxxx@musimi.dk>;tag=as06ed5480' > -- SIP/musimi-09f34188 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing Hangup("SIP/9999-09f2eb28", "") in new stack > == Spawn extension (internal, yyyyyyyy, 2) exited non-zero on > 'SIP/9999-09f2eb28' > > > I hope somebody can tell me what I'm doing wrong here. >Your sip provider is rejecting the call. This can be for many reasons. Bad user/id pass, no credit left on acct., not using proper syntax etc. Look at thier site and see how they want you to send the call to them (i.e.with the + sign before the number or maybe add or remove a 0)