[asterisk-users] Remove "name" part of SIP From header

[Warren Selby]

On Wed, May 4, 2011 at 12:10 PM, John Hablitzel <jjblitz071 at 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 at 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 at 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 at inbound:1] NoOp("SIP/2001-00000004", "Channel ID is
> SIP/2001-00000004") in new stack
> -- Executing [xxx at inbound:2] NoOp("SIP/2001-00000004", "From is <
> sip:2001 at 192.168.3.112>;tag=1c354991377") in new stack
> -- Executing [xxx at inbound:3] Set("SIP/2001-00000004",
> "CALLERID(num)=1234567890") in new stack
> -- Executing [xxx at inbound:4] Set("SIP/2001-00000004", "CALLERID(name)=")
> in new stack
> -- Executing [xxx at inbound:5] NoOp("SIP/2001-00000004", "CallerID is ""
> <1234567890>") in new stack
> -- Executing [xxx at 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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