[asterisk-dev] reset/remove SipAddHeader() headers

John Todd jtodd at digium.com
Wed Jan 14 09:07:29 CST 2009

On Jan 14, 2009, at 8:50 AM, Klaus Darilion wrote:

> Hi!
> As far as I saw in chan_sip (at least 1.4.22) once SIP headers are  
> added
> with SipAddHeader() there is no way to remove them, which is bad if  
> the
> request is rerouted to another SIp destination, e.g:
> user      Asterisk      GW1        GW2
> ---INVITE-->
>          add certain
>          header for GW1:
>          SipAddHeader(..)
>               ---INVITE--->
>               (contains additional
>               header)
>               <--500-------
>          reroute request
>          to a different GW
>              ----------INVITE-------->
>              this request also contains
>              the added SIP header
> Headers added with SipAddHeader are added to the incoming channel as
> global channel variables SIPADDHEADER01, SIPADDHEADER02, ....
> Then on outgoing requests chan_sip will traverse the channel variables
> for SIPADDHEADERxx and add them.
> Thus, it would be useful to reset the channel variables somehow, e.g.
> with an application SipAddHeaderReset();
> I could not find any existing solution, thus what do you think about
> this new application?
> thanks
> klaus

There be dragons.

I agree that there probably should be some way to do this, but a way  
to do this properly does not immediately come to mind.  For instance,  
in the event that GW2 sends a 302 redirect, how does one execute the  
SipAddHeaderReset() application or function?  That 302 redirect step  
never breaks out of Dial() - the redirect just magically happens.  The  
dialplan (or AGI, or whatever) never has a chance to execute any other  
commands such as adding headers or removing headers or whatever.  I  
don't have an answer on how this should be handled elegantly.

Even in the instance where a 500 is received and somehow the dialplan  
is re-entered, it seems burdensome to remove ALL headers and start  
from scratch.  Wouldn't you just want to remove the specific header  
that is causing problems?  This would imply some sort of indexing, or  
at the minimum, regexp-style string matching so I could say:

  exten => _X11.,SipRemoveHeader(X-PSAP-ID)

which would remove both of these headers:


Lastly, I wouldn't restrict header removal just to headers that have  
been added by prior steps in the dialplan of Asterisk.  Why not allow  
deletion of arbitrary headers?  This would permit more control over  
messaging.  (Though I will admit a glaring memory hole on my part - I  
assume Asterisk passes through headers unmodified from UA-to-UA... is  
that correct?  My first memory impression is "Yes" but the more I  
think about it the less certain I am, and I don't have time to test  
right now.)

Or am I missing your concept?


John Todd                       email:jtodd at digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083         http://www.digium.com/

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