[Asterisk-Users] Outbound calling with TDM400P

Rob Tarte rtarte at pacificcodeworks.com
Tue Feb 1 23:36:33 MST 2005


Thanks for the reply!   Unfortunately, getting rid of the g didn't 
change anything.  I'm wondering if the tones aren't getting downloaded 
correctly to the driver or something.  The driver seems to be going 
through the motions, but there is no sound.  Any other ideas?

Thanks again,

Rob

Simon Brown wrote:

>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 at lists.digium.com
>[mailto:asterisk-users-bounces at 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 at 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)
>>_______________________________________________
>>    
>>
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