<div class="gmail_quote">On Wed, May 4, 2011 at 12:10 PM, John Hablitzel <span dir="ltr"><<a href="mailto:jjblitz071@gmail.com">jjblitz071@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
Relatively new to Asterisk and SIP and am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP using Asterisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of the From header. So I want it to look like this:<br>
From: <<a href="mailto:sip%3A1234567890@192.168.3.110" target="_blank">sip:1234567890@192.168.3.110</a>>;tag=xxx<br>
<br>
However, even if I set the name to blank, using Set(CALLERID(name)= ), Asterisk always seems to put the CallerID number in the name field as well and here is what I get:<br>
From: "1234567890" <<a href="mailto:sip%3A1234567890@192.168.3.110" target="_blank">sip:1234567890@192.168.3.110</a>>;tag=xxx<br>
<br>
I cannot figure out how to get the name field to be blank. Here is the extensions.conf context that I think should work:<br>
exten => xxx,1,Noop(Channel ID is ${CHANNEL})<br>
exten => xxx,n,Noop(From is ${SIP_HEADER(From)})<br>
exten => xxx,n,Set(CALLERID(num)=1234567890)<br>
exten => xxx,n,Set(CALLERID(name)=)<br>
exten => xxx,n,Noop(CallerID is ${CALLERID(all)})<br>
exten => xxx,n(dialout),Dial(SIP/POTS1,60,o)<br>
exten => xxx,n,Hangup<br>
<br>
And my general and section from sip.conf<br>
[general]<br>
allowoverlap=no<br>
udpbindaddr=0.0.0.0<br>
tcpenable=no<br>
tcpbindaddr=0.0.0.0<br>
srvlookup=yes<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
limitonpeers=yes<br>
notifyringing=yes<br>
maxexpirery=180<br>
defaultexpirey=180<br>
<br>
[POTS1]<br>
type=friend<br>
secret=xxx<br>
context=pots_in<br>
host=dynamic<br>
dtmfmode=info<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
canreinvite=no<br>
qualify=yes<br>
call-limit=4<br>
rtptimeout=30<br>
<br>
And here is the verbose CLI output from the above configuration.<br>
-- Executing [xxx@inbound:1] NoOp("SIP/2001-00000004", "Channel ID is SIP/2001-00000004") in new stack<br>
-- Executing [xxx@inbound:2] NoOp("SIP/2001-00000004", "From is <<a href="mailto:sip%3A2001@192.168.3.112" target="_blank">sip:2001@192.168.3.112</a>>;tag=1c354991377") in new stack<br>
-- Executing [xxx@inbound:3] Set("SIP/2001-00000004", "CALLERID(num)=1234567890") in new stack<br>
-- Executing [xxx@inbound:4] Set("SIP/2001-00000004", "CALLERID(name)=") in new stack<br>
-- Executing [xxx@inbound:5] NoOp("SIP/2001-00000004", "CallerID is "" <1234567890>") in new stack<br>
-- Executing [xxx@inbound:6] Dial("SIP/2001-00000004", "SIP/POTS1,60,o") in new stack<br>
== Using SIP RTP CoS mark 5<br>
-- Called POTS1<br>
-- Got SIP response 484 "Address Incomplete" back from 192.168.3.121<br>
== Everyone is busy/congested at this time (1:0/0/1)<br clear="all"></blockquote></div><br>It doesn't look like you're ever actually sending the number you want to dial? You're setting a callerid(num), but where is the number you want to dial? What happens if you change your dial command to this:<br>
<br>exten => xxx,n(dialout),Dial(SIP/${EXTEN}@POTS1,60,o)<br><br><br>-- <br>Thanks,<br>--Warren Selby, dCAP<br><a href="http://www.selbytech.com" target="_blank">http://www.selbytech.com</a><br>