<p dir="ltr">I have attached to this Email the SIP capture taken with TCPdump. Please check it with Wireshark and let me know.</p>
<div class="gmail_quote">On 10 Jul 2015 08:28, "Podrigal, Aron" <<a href="mailto:aronp@guaranteedplus.com">aronp@guaranteedplus.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Can you provide a sip capture? </p>
<div class="gmail_quote">On Jul 10, 2015 3:04 AM, "Nabeel" <<a href="mailto:nabeelshikder@gmail.com" target="_blank">nabeelshikder@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Hi, </p>
<p dir="ltr">I am able to register my SIP clients to OpenSIPS, but when I attempt to make a call, the call does not connect to the other client. I see the call screen open and close briefly on the dialling client (approx. 1 second) and the following is seen in the log:</p>
<p dir="ltr">DBG:rr:find_first_route: No Route headers found <br>
DBG:rr:loose_route: There is no Route HF</p>
<p dir="ltr">Please advise how I can include a suitable 'route header' in my configuration. I am using TLS.</p>
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