<div dir="ltr">Hi checked your debug like.<br><br>Did you check that your SIP device ir registered with server ?<br>if yes then dial below command from CLI<br><br><b>originate sip/test02 application dial</b><br><br><br><br>
<div class="gmail_quote">On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade <span dir="ltr"><<a href="mailto:jayesh.labade@gmail.com">jayesh.labade@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi, <br><br>I am using asterisk ver 1.8.8.1.<br><br>My SIP trunk conf details are below..<br><br>[general]<br>context=default ; Default context for incoming calls<br>realm=192.168.1.55<br>allowguest=yes<br>
realmauth=yes<br>send_rpid=pai<br><br>register => <a href="mailto:test02%3Atest02@192.168.1.55" target="_blank">test02:test02@192.168.1.55</a><br><br><br>[test02]<br>type=peer<br>nat=no<br>canreinvite=no<br>host=192.168.1.55<br>
;realm=<a href="mailto:test02@192.168.1.55" target="_blank">test02@192.168.1.55</a><br>
context=incoming<br>secret=test02<br>permit=<a href="http://192.168.1.0/255.255.255.0" target="_blank">192.168.1.0/255.255.255.0</a><br>username=test02<br>fromuser=test02<br>fromdomain=192.168.1.55<br>defaultuser=test02<br>
insecure=invite,port<br>
outboundproxy=192.168.1.55<br>promiscredir=yes<br>userphone=yes<br><br>For more details you can find my paste in pastebin.. Links given below.<br><br>While Dialing call fro Xlite send following Sip header F=<a href="mailto:sip%3Atest02@192.168.1.55" target="_blank">sip:test02@192.168.1.55</a>. And if tried to register same account in asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why asterisk sends anonymous.invalid instead of domain name..Help me<div class="im">
<br clear="all">
<div><br></div><div>Best Regards,<br><b>Jayesh Labade</b><br></div>
<div>e-mail: <a href="mailto:jayesh.labade@gmail.com" target="_blank">jayesh.labade@gmail.com</a></div><br>
<br><br></div><div><div></div><div class="h5"><div class="gmail_quote">On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati <span dir="ltr"><<a href="mailto:virbhati@gmail.com" target="_blank">virbhati@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div dir="ltr">Hi,<br><br>Give the complete details about the asterisk version, and SIP trunk conf details <br><br><br><div class="gmail_quote"><div><div>On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade <span dir="ltr"><<a href="mailto:jayesh.labade@gmail.com" target="_blank">jayesh.labade@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div><div>Please help me..<br clear="all"><div><br></div><div>Best Regards,<br>
<b>Jayesh Labade</b><br></div>
<div>e-mail: <a href="mailto:jayesh.labade@gmail.com" target="_blank">jayesh.labade@gmail.com</a></div><br>
<br><br><div class="gmail_quote">On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade <span dir="ltr"><<a href="mailto:jayesh.labade@gmail.com" target="_blank">jayesh.labade@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
Hello Experts,<br><br>I have pasted my issue in <a href="http://pastebin.com/zBGVmdcY" target="_blank">http://pastebin.com/zBGVmdcY</a><br><span style="font-weight:normal"><font><font color="#204a87"></font></font></span><br>
I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to authenticate on INVITE to '"Anonymous" <sip:test02@anonymous.invalid>;tag=as57d3a806'<br>
<span style="font-weight:normal"><font><font color="#204a87"></font></font></span><span style="font-weight:bold;color:rgb(32,74,135)"></span>i am unable to make outbound call from this trunk. while if i registered this trunk in softphone like Xlite, there is no problem with outbound calls. Help me.<br>
<span style="font-weight:normal"><font><font color="#204a87"></font></font></span><br>please find sip.conf file in <a href="http://pastebin.com/zBGVmdcY" target="_blank">http://pastebin.com/zBGVmdcY</a><br><br>I have pasted sip debug with verbosity of failed call <a href="http://pastebin.com/jL2ki0s8" target="_blank">http://pastebin.com/jL2ki0s8</a><br>
<br clear="all"><div><br></div><div>Best Regards,<span><font color="#888888"><br><b>Jayesh Labade</b><br></font></span></div><span><font color="#888888">
<div>e-mail: <a href="mailto:jayesh.labade@gmail.com" target="_blank">jayesh.labade@gmail.com</a></div><br>
</font></span></blockquote></div><br>
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<br clear="all"><br>-- <br><div dir="ltr"><br>Thanks and regards<br>
<br> Virendra Bhati<br><a href="tel:%2B91-8885268942" value="+918885268942" target="_blank">+91-8885268942</a><br>Software Engineer<br></div><br>
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New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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<br> Virendra Bhati<br>+91-8885268942<br>Software Engineer<br></div><br>
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