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 _______________________________________________ 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