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<DIV><FONT face="Comic Sans MS" size=2>Hi Drew. Thanks for the tips. My Line 1
works as I'd like it to, and I could be wrong, but I don't think changing the
dialplan there will help. I really just want to be able to dial local phone
calls (7 digits) and have it go out the SPA3102, without having to dial twice.
This is a snip what I have so far.</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2>extentions.conf<BR>exten =>
_NXXXXXX,1,Dial(SIP/201/${EXTEN},20)<BR>exten =>
_NXXXXXX,2,Hangup<BR><BR>sip.conf<BR>[201]<BR>type=friend<BR>username=x<BR>secret=x<BR>host=dynamic<BR>context=sip<BR>nat=yes<BR>canreinvite=yes<BR>qualify=yes<BR>subscribecontext=localextensions<BR>dtmfmode=rfc2833<BR>vmexten=voicemail<BR>disallow=all<BR>allow=ulaw<BR>allow=gsm<BR><BR>On
the SPA (in the "PSTN Line" tab)<BR>Dial Plan 1:
(<xxxxxxxS0:@gw0>)<BR>Dial Plan 2: S0<:255><BR><BR>DialPlan 1
is just what I have for now<BR>DialPlan 2 is my extention 255, with a PSTN call
comes in it rings my SIP phone.<BR></FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2>I set my VoIP Answer Delay to 0 (from 1,
I also tried 5 some time ago) and I set the SPA To PSTN Gain to 5 and now
15.</FONT></DIV>
<DIV><FONT face="Comic Sans MS" size=2></FONT> </DIV>
<DIV><FONT face="Comic Sans MS" size=2><FONT face=Arial>With things the way I
have them now, when I dial a local number, I get a single DTMF tone on the
phoneline, not sure what digit it is.</FONT></DIV>
<DIV><BR>Tim<BR></DIV></FONT>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=drew@oanda.com href="mailto:drew@oanda.com">Drew Gibson</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, August 07, 2007 5:55
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [asterisk-users] Outbound
dialing</DIV>
<DIV><BR></DIV>Tim,<BR><BR>If the Asterisk stuff below doesn't fix it, try the
docs at <A class=moz-txt-link-freetext
href="http://www.jmgtechnology.com.au/spa_3000_guide.pdf">http://www.jmgtechnology.com.au/spa_3000_guide.pdf</A><BR><BR>Ensure
you enable VoIP to PSTN gateway mode and that "PSTN Line" is registered with
Asterisk. This is probably OK as you appear to get dialtone back from the SPA.
If you are calling from the phone on "Line 1", make all calls go through
Asterisk. See above docs for details.<BR><BR>In case you are dialing from a
phone on "Line 1", here is the "Line 1" dialplan from my home
SPA3102...<BR><BR>(*xx|[3469]11|0|00|[29]xxxxxxxxx|1xxx[2-9]xxxxxx|2[01]x|50[01]|xxxxxxxxxxxx.)<BR><BR>I
can't remember if that is default or if I tweaked it. Works in
Ontario.<BR><BR>If that is OK, try increasing the gain "SPA to PSTN". If the
gain is too low, the DTMF may not be recognised by the CO. I found this out
whilst troubleshooting echo
problems.<BR><BR>regards,<BR><BR>Drew<BR><BR><BR>Nicholas Blasgen wrote:
<BLOCKQUOTE
cite=mid:272f26b80708071257w431f2000uf568af0a8807027f@mail.gmail.com
type="cite">
<DIV>Not specific to the SPA3102, but just normal outbound dialing is as
follows:</DIV>
<DIV> </DIV>
<DIV>exten => _1NXXNXXXXXX,1,Dial(<trunk
type>/<name>/${EXTEN})<BR> </DIV>
<DIV>or if you want to require people to dial 9, then:</DIV>
<DIV> </DIV>
<DIV>exten => _91NXXNXXXXXX,1,Dial(<trunk
type>/<name>/${EXTEN})<BR> </DIV>
<DIV>or if you're like me and you're used to a cell phone and don't like
dialing the 1:</DIV>
<DIV> </DIV>
<DIV>exten => _NXXNXXXXXX,1,Dial(<trunk
type>/<name>/1${EXTEN})</DIV>
<DIV><BR> </DIV>
<DIV><SPAN class=gmail_quote>On 8/7/07, <B class=gmail_sendername>Tim
Johnson</B> <<A href="mailto:tjapml@cometonovascotia.ca"
moz-do-not-send="true">tjapml@cometonovascotia.ca</A>> wrote:</SPAN>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">Hello
all. I am just getting back into Asterisk and I am setting up
my<BR>Linksys SPA3102. I have incoming calls working fine, as is the phone
<BR>plugged into the unit. My problem is I cannot get the SPA3102 to
dial<BR>a phone number automatically. I can call the extention of the PSTN
and<BR>I get a second dialtone, and I can then manually dial. I'd like to
be <BR>able to have Asterisk pass the number I dialed to the SPA and have
it<BR>dialout. I've played with dialplans on the SPA I've found during
my<BR>googling, but I think it might be something I am missing in
my<BR>extentions.conf file. Any
ideas?<BR><BR>Tim<BR><BR>_______________________________________________<BR>--Bandwidth
and Colocation Provided by <A href="http://www.api-digital.com--"
moz-do-not-send="true">http://www.api-digital.com--</A><BR><BR>asterisk-users
mailing list <BR>To UNSUBSCRIBE or update options visit:<BR> <A
href="http://lists.digium.com/mailman/listinfo/asterisk-users"
moz-do-not-send="true">http://lists.digium.com/mailman/listinfo/asterisk-users</A><BR></BLOCKQUOTE></DIV><BR><BR
clear=all><BR>-- <BR>/Nick <PRE wrap=""><HR width="90%" SIZE=4>
_______________________________________________
--Bandwidth and Colocation Provided by <A class=moz-txt-link-freetext href="http://www.api-digital.com">http://www.api-digital.com</A>--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<A class=moz-txt-link-freetext href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></PRE></BLOCKQUOTE><BR><PRE class=moz-signature cols="72">--
Drew Gibson
Systems Administrator
OANDA Corporation
416-593-6767 x322
<A class=moz-txt-link-abbreviated href="http://www.oanda.com">www.oanda.com</A>
</PRE>
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