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@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@yahoo.com] Sent: Sunday, February 06, 2005 11:26 AM To: serusers@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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