asterisk users - Jun 2010 - Caller id, sip header from problem

Hello all,

My pbx server is connected to a sip gateway, when I call an originate
command from the asterisk console, to establish a sip connection, the
gateway doesn't accept URL with white spaces, for example:

       * Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport *

*      From: "PBX SERVER" <sip:PBX SERVER at
10.10.1.10>;tag=as2512881b *

*      To: <sip:927817839 at 10.10.1.250:5060>;tag=2615730116
*

*      Contact: <sip:PBX SERVER at 10.10.1.10>
*

*      Call-ID: 454df9c904486e7647231af102a05b34 at 10.10.1.10 *

*      CSeq: 102 ACK*

*      Max-Forwards: 70*


The sip gateway will respond with the following message:


        *SIP/2.0 400 Bad Request *

*      Via: SIP/2.0/UDP 10.10.1.10:5060;branch=z9hG4bK387d772e;rport *

*      From: "PBX SERVER" <sip:PBX SERVER at
10.10.1.10>;tag=as2512881b *

*      To: <sip:927817839 at 10.10.1.250:5060>;tag=2615730116 *

*      Call-ID: 454df9c904486e7647231af102a05b34 at 10.10.1.10 *

*      CSeq: 102 INVITE
*
*      Content-Type: text/plain *

*      Content-Length: 23 *



The "PBX SERVER" name is set in the sip.conf in the callerid
parameter.

Question:

Is it possible, without trimming the callerid parameter, to set some type of
variable in asterisk to trim automatically.

Thanks in advance,

Alex
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