<span style="font-family: arial narrow,sans-serif;">Hello all, </span><br style="font-family: arial narrow,sans-serif;"><br style="font-family: arial narrow,sans-serif;"><span style="font-family: arial narrow,sans-serif;">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:</span><br style="font-family: arial narrow,sans-serif;">
<br style="font-family: arial narrow,sans-serif;">
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<p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"> <i> Via: SIP/2.0/UDP
10.10.1.10:5060;branch=z9hG4bK387d772e;rport </i>
</p>
<p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><i>
From: "PBX SERVER" <sip:<b>PBX SERVER</b>@<a href="http://10.10.1.10">10.10.1.10</a>>;tag=as2512881b </i>
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<p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><i>
To:
<<a href="http://sip:927817839@10.10.1.250:5060">sip:927817839@10.10.1.250:5060</a>>;tag=2615730116<br></i> </p>
<p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><i>
Contact: <sip:PBX <a href="mailto:SERVER@10.10.1.10">SERVER@10.10.1.10</a>><br></i> </p>
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Call-ID:
<a href="mailto:454df9c904486e7647231af102a05b34@10.10.1.10">454df9c904486e7647231af102a05b34@10.10.1.10</a> </i>
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<p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><i>
CSeq: 102 ACK</i></p><p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><i>
Max-Forwards: 70</i></p><p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><br></p><p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT">
The sip gateway will respond with the following message:</p><p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT"><br></p><p style="margin-bottom: 0in; font-weight: normal; font-family: arial narrow,sans-serif;" lang="pt-PT" align="LEFT">
<meta http-equiv="CONTENT-TYPE" content="text/html; charset=utf-8">
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<i>SIP/2.0 400 Bad Request </i>
</p>
<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i> Via: SIP/2.0/UDP
10.10.1.10:5060;branch=z9hG4bK387d772e;rport </i>
</p>
<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i>
From: "PBX SERVER" <sip:PBX <a href="mailto:SERVER@10.10.1.10">SERVER@10.10.1.10</a>>;tag=as2512881b </i>
</p>
<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i>
To:
<<a href="http://sip:927817839@10.10.1.250:5060">sip:927817839@10.10.1.250:5060</a>>;tag=2615730116 </i>
</p>
<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i> Call-ID:
<a href="mailto:454df9c904486e7647231af102a05b34@10.10.1.10">454df9c904486e7647231af102a05b34@10.10.1.10</a> </i>
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<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i>
CSeq: 102 INVITE<br></i> </p><i> Content-Type: text/plain </i>
<p style="margin-bottom: 0in; font-weight: normal;" lang="pt-PT" align="LEFT"><i>
Content-Length: 23 </i>
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<p></p>
<br><br>The "PBX SERVER" name is set in the sip.conf in the callerid parameter.<br><br>Question: <br><br>Is it possible, without trimming the callerid parameter, to set some type of variable in asterisk to trim automatically. <br>
<br>Thanks in advance,<br><br>Alex <br>