I am trying to place an analog outbound call from a Sipura SPA-841
through a * server with a TDM400P and 4 FXO's. When I call in from an
analog line everything works fine, I can talk over the SIP phone. When
I call out, * says:
== Spawn extension (from-sip, [phonenumber], 1) exited non-zero on
'SIP/sipphone-d29d'
-- Executing Dial("SIP/sipphone-9eb0",
"Zap/g1/[phonenumber]|60") in new
stack
-- Called g1/[phonenumber]
-- Zap/1-1 answered SIP/sipphone-9eb0
And then I get silence. The phone doesn't ring on the other end. I
have attached my configuration files.
Any help would be greatly appreciated,
Rob
------------------------------------- sip.conf ----------------------------
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
[sipphone]
type=friend
context=from-sip
username=sipphone
fromuser=sipphone
callerid=Incoming Call<101>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
incominglimit=1
mailbox=101@default
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729
-------------------------------- zaptel.conf -----------------------
loadzone = us
defaultzone=us
fxsks=1-4
-------------------------------- zapata.conf -----------------------
[channels]
switchtype=national
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
callerid=asreceived
group=1
signalling=fxs_ks
languange=en
context=default
channel => 1-4
-------------------------------- extensions.conf -----------------------
[general]
static=yes
writeprotect=no
[globals]
IAXINFO=guest ; IAXtel username/password
OUTGOING => Zap/1
[from-sip]
ignorepat => 9
exten => _X.,1,Dial(Zap/g1/${EXTEN},60)
exten => _X.,2,Hangup
;exten => _NXXXXXXX,1,Dial(Zap/g1)
[default]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,Dial(SIP/sipphone)
A little more investigation: I hooked up another phone to a splitter so I could listen to the outbound line. There are no sounds of any sort coming out on the line when the FXO should be dialing. I put some debug in the zaptel driver, and I can see the driver trying to dial. It calls __do_dtmf() with all of the digits that I would like it to dial, but there is no sound on the wire. Any ideas? Thanks, Rob Rob Tarte wrote:> I am trying to place an analog outbound call from a Sipura SPA-841 > through a * server with a TDM400P and 4 FXO's. When I call in from an > analog line everything works fine, I can talk over the SIP phone. > When I call out, * says: > > == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on > 'SIP/sipphone-d29d' > -- Executing Dial("SIP/sipphone-9eb0", "Zap/g1/[phonenumber]|60") in > new stack > -- Called g1/[phonenumber] > -- Zap/1-1 answered SIP/sipphone-9eb0 > > And then I get silence. The phone doesn't ring on the other end. I > have attached my configuration files. > > Any help would be greatly appreciated, > > Rob > > ------------------------------------- sip.conf > ---------------------------- > [general] > context=default > port=5060 > bindaddr=0.0.0.0 > srvlookup=yes > > > [sipphone] > type=friend > context=from-sip > username=sipphone > fromuser=sipphone > callerid=Incoming Call<101> > host=dynamic > nat=no > canreinvite=yes > dtmfmode=info > incominglimit=1 > > > mailbox=101@default > disallow=all > allow=ulaw > > > allow=alaw > allow=g723.1 > allow=g729 > > > -------------------------------- zaptel.conf ----------------------- > loadzone = us > defaultzone=us > fxsks=1-4 > > > -------------------------------- zapata.conf ----------------------- > > > [channels] > switchtype=national > rxwink=300 ; Atlas seems to use long (250ms) winks > usecallerid=yes > hidecallerid=no > callwaiting=yes > usecallingpres=yes > callwaitingcallerid=yes > threewaycalling=yes > transfer=yes > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=yes > rxgain=0.0 > txgain=0.0 > callgroup=1 > pickupgroup=1 > immediate=no > callerid=asreceived > > > group=1 > signalling=fxs_ks > languange=en > context=default > channel => 1-4 > > > -------------------------------- extensions.conf ----------------------- > [general] > static=yes > writeprotect=no > > > [globals] > IAXINFO=guest ; IAXtel > username/password > OUTGOING => Zap/1 > > > [from-sip] > ignorepat => 9 > exten => _X.,1,Dial(Zap/g1/${EXTEN},60) > exten => _X.,2,Hangup > ;exten => _NXXXXXXX,1,Dial(Zap/g1) > > > [default] > exten => s,1,Wait,1 ; Wait a second, just for fun > exten => s,2,Answer ; Answer the line > exten => s,3,Dial(SIP/sipphone) > _______________________________________________ > 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-- Robert Tarte Pacific CodeWorks 1347 Pacific Ave., Suite 202 Santa Cruz, CA 95060 (p) 831-426-7582 (f) 831-426-7584
Try removing the g from the dial command:
exten => _X.,1,Dial(Zap/1/${EXTEN},60)
exten => _X.,2,Hangup ;
exten => _NXXXXXXX,1,Dial(Zap/1)
Simon Brown
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rob Tarte
Sent: Wednesday, 2 February 2005 16:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Outbound calling with TDM400P
A little more investigation:
I hooked up another phone to a splitter so I could listen to the outbound
line. There are no sounds of any sort coming out on the line when the FXO
should be dialing. I put some debug in the zaptel driver, and I can see the
driver trying to dial. It calls __do_dtmf() with all of the digits that I
would like it to dial, but there is no sound on the wire. Any ideas?
Thanks,
Rob
Rob Tarte wrote:
> I am trying to place an analog outbound call from a Sipura SPA-841
> through a * server with a TDM400P and 4 FXO's. When I call in from an
> analog line everything works fine, I can talk over the SIP phone.
> When I call out, * says:
>
> == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on
> 'SIP/sipphone-d29d'
> -- Executing Dial("SIP/sipphone-9eb0",
"Zap/g1/[phonenumber]|60") in
> new stack
> -- Called g1/[phonenumber]
> -- Zap/1-1 answered SIP/sipphone-9eb0
>
> And then I get silence. The phone doesn't ring on the other end. I
> have attached my configuration files.
>
> Any help would be greatly appreciated,
>
> Rob
>
> ------------------------------------- sip.conf
> ----------------------------
> [general]
> context=default
> port=5060
> bindaddr=0.0.0.0
> srvlookup=yes
>
>
> [sipphone]
> type=friend
> context=from-sip
> username=sipphone
> fromuser=sipphone
> callerid=Incoming Call<101>
> host=dynamic
> nat=no
> canreinvite=yes
> dtmfmode=info
> incominglimit=1
>
>
> mailbox=101@default
> disallow=all
> allow=ulaw
>
>
> allow=alaw
> allow=g723.1
> allow=g729
>
>
> -------------------------------- zaptel.conf -----------------------
> loadzone = us defaultzone=us
> fxsks=1-4
>
>
> -------------------------------- zapata.conf -----------------------
>
>
> [channels]
> switchtype=national
> rxwink=300 ; Atlas seems to use long (250ms) winks
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> callerid=asreceived
>
>
> group=1
> signalling=fxs_ks
> languange=en
> context=default
> channel => 1-4
>
>
> -------------------------------- extensions.conf
> ----------------------- [general] static=yes writeprotect=no
>
>
> [globals]
> IAXINFO=guest ; IAXtel
> username/password
> OUTGOING => Zap/1
>
>
> [from-sip]
> ignorepat => 9
> exten => _X.,1,Dial(Zap/g1/${EXTEN},60) exten => _X.,2,Hangup ;exten
> => _NXXXXXXX,1,Dial(Zap/g1)
>
>
> [default]
> exten => s,1,Wait,1 ; Wait a second, just for fun
> exten => s,2,Answer ; Answer the line
> exten => s,3,Dial(SIP/sipphone)
> _______________________________________________
> 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
--
Robert Tarte
Pacific CodeWorks
1347 Pacific Ave., Suite 202
Santa Cruz, CA 95060
(p) 831-426-7582
(f) 831-426-7584
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