Remove CallerID Name from sip invite

I am trying to run a proof of concept using Asterisk to make an outbound call through an Audiocodes gateway via SIP. The specific requirements of the gateway in this configuration specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field part of the From header. So it should look like this:
From: sip:1234567890@192.168.3.110;tag=xxx

However what I get from Asterisk is this:
From: “1234567890” sip:1234567890@192.168.3.110;tag=xxx

I cannot figure out how to get the name field to be blank. Here is my very simple extension from extensions.conf:
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.
– 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;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)

As you can see the Noop on the Caller ID shows that the name is blank, but Asterisk seems to default somehow to putting the number in the name field if it is blank when the Invite is created. Is there a way to configure Asterisk not to do this? Asterisk version is 1.6.1.12.

Thanks in advance for any insight you can provide.

Same problem here, any help please?

Use a SIP compliant gateway!

It is not of sufficient interest to me to look at the code, but my guess is that the channel driver is adding back the user friendly part of the From: line, as a default, so you may need to modify the source code. That should be very easy, as long as you don’t mind suppressing it for internal phones, as well.