<div dir="ltr">Hi Kevin,<div><br></div><div>unless I've stuffed something up... it's not working..</div><div><br></div><div style>Have set user_cli as well as the E164 as well. From the sip trace, I see that from the SPCE to the provider, the INVITE is as follows...</div>
<div style><br></div><div style>123.123.123.123 is the SPCE, 213.213.213.213 the provider.</div><div style><br></div><div style>As you can see, somehow the From: header has 9030 which is the extension number of an Asterisk system. Therefore 9030 is not a valid number and thus the provider puts it as anonymous. I can't seem to force the CLI.</div>
<div style><br></div><div style>Any ideas what's happening?</div><div style><br></div><div style><div><div>U 2013/05/10 01:05:03.335037 <a href="http://123.123.123.123:5060">123.123.123.123:5060</a> -> <a href="http://213.213.213.213:5060">213.213.213.213:5060</a></div>
<div>INVITE sip:123456789@213.213.213.213:5060;transport=udp SIP/2.0.</div><div>Max-Forwards: 10.</div><div>Record-Route: <sip:123.123.123.123;r2=on;lr=on;ftag=3686E06C-518BBB1F000519FA-062EA700;ngcplb=yes>.</div><div>
Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=3686E06C-518BBB1F000519FA-062EA700;ngcplb=yes>.</div><div>Via: SIP/2.0/UDP 123.123.123.123;branch=z9hG4bK1bf6.45addc0c691ddcb65ab53185af6baca1.0.</div><div>Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKOqRJLadA;rport=5080.</div>
<div>From: <<a href="mailto:sip%3A9030@123.123.123.123">sip:9030@123.123.123.123</a>>;tag=3686E06C-518BBB1F000519FA-062EA700.</div><div>To: <<a href="mailto:sip%3A123456789@213.213.213.213">sip:123456789@213.213.213.213</a>>.</div>
<div>CSeq: 10 INVITE.</div><div>Call-ID: 243cdb514476fa870cabdef15e9c406f@123.123.123.123_b2b-1.</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.</div><div>Supported: replaces.</div><div>
P-Asserted-Identity: <<a href="mailto:sip%3A9030@123.123.123.123">sip:9030@123.123.123.123</a>>.</div><div>Content-Type: application/sdp.</div><div>Content-Length: 252.</div><div>Contact: <sip:ngcp-lb@123.123.123.123:5060;ngcpct='sip:<a href="http://127.0.0.1:5080">127.0.0.1:5080</a>'>.</div>
<div>.</div><div>v=0.</div><div>o=root 13171 13172 IN IP4 123.123.123.123.</div><div>s=session.</div><div>c=IN IP4 123.123.123.123.</div><div>t=0 0.</div><div>m=audio 32354 RTP/AVP 18 101.</div><div>a=rtpmap:18 G729/8000.</div>
<div>a=fmtp:18 annexb=no.</div><div>a=rtpmap:101 telephone-event/8000.</div><div>a=fmtp:101 0-16.</div><div>a=ptime:20.</div><div>a=sendrecv.</div><div>a=rtcp:32355.</div></div><div><br></div></div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Fri, May 10, 2013 at 12:21 AM, Kevin Masse <span dir="ltr"><<a href="mailto:kmasse@questblue.com" target="_blank">kmasse@questblue.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Good morning, if you are looking to force the outbound caller ID of the subscriber make sure you have the DID in the e.164 number field.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">That should force the outbound caller ID to be that number.<u></u><u></u></span></p><p class="MsoNormal">
<span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">For clients that have multiple outbound caller ID leave out the e.164 subscriber number and place all the numbers in the alias area instead.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Kevin<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:spce-user-bounces@lists.sipwise.com" target="_blank">spce-user-bounces@lists.sipwise.com</a> [mailto:<a href="mailto:spce-user-bounces@lists.sipwise.com" target="_blank">spce-user-bounces@lists.sipwise.com</a>] <b>On Behalf Of </b>Barry Flanagan<br>
<b>Sent:</b> Thursday, May 09, 2013 10:18 AM<br><b>To:</b> Martin Wong<br><b>Cc:</b> <a href="mailto:spce-user@lists.sipwise.com" target="_blank">spce-user@lists.sipwise.com</a><br><b>Subject:</b> Re: [Spce-user] Force CLI<u></u><u></u></span></p>
<div><div class="h5"><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">On 9 May 2013 13:30, Martin Wong <<a href="mailto:martin.wong@binaryelements.com.au" target="_blank">martin.wong@binaryelements.com.au</a>> wrote:<u></u><u></u></p>
<div><div><blockquote style="border:none;border-left:solid #cccccc 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in"><div><p class="MsoNormal">Hi,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p>
</div><div><p class="MsoNormal">I am testing a few subscribers. <u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">One is from a normal sip phone ... the <a href="mailto:fromuser@xx.xx.xx.xx" target="_blank">fromuser@xx.xx.xx.xx</a> is the correct CLI set, therefore the caller ID is correctly displayed at the end point device as it goes through the downstream provider.<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">The other is an asterisk box. Even though the CLI is set in the SPCE portal, it comes up as something else in the <a href="mailto:fromuser@xx.xx.xx.xx" target="_blank">fromuser@xx.xx.xx.xx</a><u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I would like to force the CLI no matter what the asterisk box is giving out.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p>
</div></div></blockquote><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Hi <u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">I think you want to set user_cli in the subscriber's preferences:<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">user_cli: SIP username (the localpart of the whole SIP URI, eg., "user" of SIP URI "<a href="mailto:user@example.com" target="_blank">user@example.com</a>"). "user-provided calling line identification" - specifies the SIP username that is used for outgoing calls. If set, this is put in the SIP "From" header (as user-provided calling number) if a client sends a CLI which is not allowed by "allowed_clis" or if "allowed_clis" is not set.<u></u><u></u></p>
</div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Hope this helps.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div>
<div><p class="MsoNormal">-Barry<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div></div></div></div></div></div></div></div></blockquote></div><br></div></div>