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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?<br>
<br>
Thanks again,<br>
<br>
Rob<br>
<br>
Simon Brown wrote:
<blockquote
cite="mid6D9F7447ED9582449D6385885F3B021E095980@astronomix.sydney.otterson"
type="cite">
<pre wrap="">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: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] 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:
</pre>
<blockquote type="cite">
<pre wrap="">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
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">[sipphone]
type=friend
context=from-sip
username=sipphone
fromuser=sipphone
callerid=Incoming Call<101>
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
incominglimit=1
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">mailbox=101@default
disallow=all
allow=ulaw
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">allow=alaw
allow=g723.1
allow=g729
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">-------------------------------- zaptel.conf -----------------------
loadzone = us defaultzone=us
fxsks=1-4
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">-------------------------------- zapata.conf -----------------------
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">[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
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">group=1
signalling=fxs_ks
languange=en
context=default
channel => 1-4
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">-------------------------------- extensions.conf
----------------------- [general] static=yes writeprotect=no
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">[globals]
IAXINFO=guest ; IAXtel
username/password
OUTGOING => Zap/1
</pre>
</blockquote>
<pre wrap=""><!---->
</pre>
<blockquote type="cite">
<pre wrap="">[from-sip]
ignorepat => 9
exten => _X.,1,Dial(Zap/g1/${EXTEN},60) exten => _X.,2,Hangup ;exten
=> _NXXXXXXX,1,Dial(Zap/g1)
</pre>
</blockquote>
<blockquote type="cite">
<pre wrap="">[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)
_______________________________________________
</pre>
</blockquote>
</blockquote>
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