On 5/4/2011 4:04 PM, Warren Selby wrote:
On Wed, May 4, 2011 at 12:10 PM, John Hablitzel <jjblitz...@gmail.com
<mailto:jjblitz...@gmail.com>> wrote:
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:
From: <sip:1234567890@192.168.3.110
<mailto:sip%3A1234567890@192.168.3.110>>;tag=xxx
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:
From: "1234567890" <sip:1234567890@192.168.3.110
<mailto:sip%3A1234567890@192.168.3.110>>;tag=xxx
I cannot figure out how to get the name field to be blank. Here is
the extensions.conf context that I think should work:
exten => xxx,1,Noop(Channel ID is ${CHANNEL})
exten => xxx,n,Noop(From is ${SIP_HEADER(From)})
exten => xxx,n,Set(CALLERID(num)=1234567890)
exten => xxx,n,Set(CALLERID(name)=)
exten => xxx,n,Noop(CallerID is ${CALLERID(all)})
exten => xxx,n(dialout),Dial(SIP/POTS1,60,o)
exten => xxx,n,Hangup
And my general and section from sip.conf
[general]
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
limitonpeers=yes
notifyringing=yes
maxexpirery=180
defaultexpirey=180
[POTS1]
type=friend
secret=xxx
context=pots_in
host=dynamic
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
qualify=yes
call-limit=4
rtptimeout=30
And here is the verbose CLI output from the above configuration.
-- Executing [xxx@inbound:1] NoOp("SIP/2001-00000004", "Channel ID
is SIP/2001-00000004") in new stack
-- Executing [xxx@inbound:2] NoOp("SIP/2001-00000004", "From is
<sip:2001@192.168.3.112
<mailto:sip%3A2001@192.168.3.112>>;tag=1c354991377") in new stack
-- Executing [xxx@inbound:3] Set("SIP/2001-00000004",
"CALLERID(num)=1234567890") in new stack
-- Executing [xxx@inbound:4] Set("SIP/2001-00000004",
"CALLERID(name)=") in new stack
-- Executing [xxx@inbound:5] NoOp("SIP/2001-00000004", "CallerID
is "" <1234567890>") in new stack
-- Executing [xxx@inbound:6] Dial("SIP/2001-00000004",
"SIP/POTS1,60,o") in new stack
== Using SIP RTP CoS mark 5
-- Called POTS1
-- Got SIP response 484 "Address Incomplete" back from 192.168.3.121
== Everyone is busy/congested at this time (1:0/0/1)
It doesn't look like you're ever actually sending the number you want
to dial? You're setting a callerid(num), but where is the number you
want to dial? What happens if you change your dial command to this:
exten => xxx,n(dialout),Dial(SIP/${EXTEN}@POTS1,60,o)
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
--
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I tried your dial command and it fails as well. This is a non-standard
type of configuration on the gateway used for making outbound CAMA type
of calls with DID wink and MF signalling. All I have to do is an Invite
to the system with the From header as described above and the gateway
will pull the information it needs from the header. I can make it work
in one mode where it is expecting information in both parts (name and
number), but it fails in another mode where it just wants the number.
--
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