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 [email protected]>;tag=as2512881b *
* To: <sip:[email protected]:5060>;tag=2615730116
*
* Contact: <sip:PBX [email protected]>
*
* Call-ID: [email protected] *
* 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 [email protected]>;tag=as2512881b *
* To: <sip:[email protected]:5060>;tag=2615730116 *
* Call-ID: [email protected] *
* 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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