[Serusers] Modifying "From" header field of an INVITE message

[Greger V. Teigre]

Sorry about coming up with this now, I'm normally off the list in the 
weekends ;-)

Without saying anything about whether you *should* manipulate From, it is 
possible.  Here is something that should do what you originally wanted:
subst("/^From:."(.*)sip:([^@]*@[a-zA-Z0-9.]+(.*))$/From: "+\1sip:+\2/");

If you just changed the sip address and not the "name" tag, it would be a 
safer replacement:
subst("/^From:(.*)sip:([^@]*@[a-zA-Z0-9.]+(.*))$/From:\1sip:+\2/");

g-)

scm-j at nuntius.com wrote:
> Thanks Girish, Greg, John and Andres. I made the change to the
> Asterisk dialplan and the "From" field is now appropriately presented
> to SER. Andres, I could not use re-invite as my application requires
> Asterisk to bridge the RTP media path. Thanks a lot!!!
>
> Another problem I face is when I call the toll-free number (that
> points to my SER) from a phone with the caller ID blocked, although I
> receive the ANI (from Level-3) and is passed onto Asterisk (also
> shows up in $CALLERID), when Asterisk makes the outbound call (Dial
> command), it changes the "From" field to "Unknown". However, if I
> make the same call with the callerID unblocked, the ANI is present in
> the "From" field.
>
> Look like there is some kind of privacy field in the SIP (or SDP)
> header that triggers Asterisk to block the callerID on calls
> originating from blocked numbers. The problem I face due to this, is
> that I am unable to terminate calls to other toll-free numbers, as
> they require the ANI. Any clues?
>
> regards,
> SCM
>
> -----Original Message-----
> From: Andrés Parra L. [mailto:apl_1980b at yahoo.com]
> Sent: Sunday, February 06, 2005 11:26 AM
> To: serusers at lists.iptel.org
> Subject: Re: [Serusers] Modifying "From" header field of an INVITE
> message
>
> Don't hesitate anymore, use asterisk to do so with the
> SetCallerID command in extensions.conf. You should
> forward the call through Asterisk from SER. If you
> don't want to have all the media pass through you, and
> i beleive you don't, use canreinvite=yes in sip.conf,
> you should fight a little bit with the codecs with
> allow and disallow but it should be fine. The next
> links could help you with the reinvitation issue:
> http://www.voip-info.org/wiki-Asterisk+SIP+media+path
> http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
>
>
> Hope it helps.
>
> Andres
>
>
>
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