<div dir="ltr"><div><div><div>Hi,<br><br></div>This is the log. ex dialling 0 from snom phone<br><br><br><--- Received SIP request (1230 bytes) from UDP:<a href="http://123.231.72.210:33878">123.231.72.210:33878</a> ---><br>INVITE <a href="mailto:sip%3A0@54.206.59.252">sip:0@54.206.59.252</a>;user=phone SIP/2.0<br>Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport<br>From: "outburns00-nhvg5vjjn6-2001" <<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252">sip:outburns00-nhvg5vjjn6-2001@54.206.59.252</a>>;tag=1bb809zgaa<br>To: <<a href="mailto:sip%3A0@54.206.59.252">sip:0@54.206.59.252</a>;user=phone><br>Call-ID: 313437333433383639323238313539-ahn3begiq66q<br>CSeq: 1 INVITE<br>Max-Forwards: 70<br>User-Agent: snom710/<a href="http://8.7.5.35">8.7.5.35</a><br>Contact: <<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835">sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835</a>>;reg-id=1<br>X-Serialnumber: 000413747C96<br>P-Key-Flags: resolution="31x13", keys="4"<br>Accept: application/sdp<br>Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE<br>Allow-Events: talk, hold, refer, call-info<br>Supported: timer, 100rel, replaces, from-change<br>Session-Expires: 3600<br>Min-SE: 90<br>Content-Type: application/sdp<br>Content-Length: 405<br><br>v=0<br>o=root 2136927789 2136927789 IN IP4 192.168.2.28<br>s=call<br>c=IN IP4 123.231.72.210<br>t=0 0<br>m=audio 62724 RTP/AVP 9 0 8 3 99 112 18 101<br>a=rtpmap:9 G722/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:99 G726-32/8000<br>a=rtpmap:112 AAL2-G726-32/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-15<br>a=ptime:20<br>a=sendrecv<br><br><--- Transmitting SIP response (572 bytes) to UDP:<a href="http://123.231.72.210:33878">123.231.72.210:33878</a> ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 123.231.72.210:45835;rport=33878;received=123.231.72.210;branch=z9hG4bK-bskkkx1t5bas<br>Call-ID: 313437333433383639323238313539-ahn3begiq66q<br>From: "outburns00-nhvg5vjjn6-2001" <<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252">sip:outburns00-nhvg5vjjn6-2001@54.206.59.252</a>>;tag=1bb809zgaa<br>To: <<a href="mailto:sip%3A0@54.206.59.252">sip:0@54.206.59.252</a>;user=phone>;tag=z9hG4bK-bskkkx1t5bas<br>CSeq: 1 INVITE<br>WWW-Authenticate: Digest  realm="asterisk",nonce="1473438693/ef923d25464dbedc1dbd85e0ccea08b7",opaque="210b270d7abb2354",algorithm=md5,qop="auth"<br>Server: Asterisk PBX certified/13.8-cert2<br>Content-Length:  0<br><br><br><--- Received SIP request (487 bytes) from UDP:<a href="http://123.231.72.210:33878">123.231.72.210:33878</a> ---><br>ACK <a href="mailto:sip%3A0@54.206.59.252">sip:0@54.206.59.252</a>;user=phone SIP/2.0<br>Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport<br>From: "outburns00-nhvg5vjjn6-2001" <<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252">sip:outburns00-nhvg5vjjn6-2001@54.206.59.252</a>>;tag=1bb809zgaa<br>To: <<a href="mailto:sip%3A0@54.206.59.252">sip:0@54.206.59.252</a>;user=phone>;tag=z9hG4bK-bskkkx1t5bas<br>Call-ID: 313437333433383639323238313539-ahn3begiq66q<br>CSeq: 1 ACK<br>Max-Forwards: 70<br>User-Agent: snom710/<a href="http://8.7.5.35">8.7.5.35</a><br>Contact: <<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835">sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835</a>>;reg-id=1<br>Content-Length: 0<br><br><br></div>Best Regards,<br></div>Madushan<br><div><div><br><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga <span dir="ltr"><<a href="mailto:mgliyanage.rc@gmail.com" target="_blank">mgliyanage.rc@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div>Hi,<br><br></div>I'm trying to setup snom 710 phone with 
asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i
 don't hear anything on the phone and asterisk CLI also does not show 
anything. my config is. please advice.<br><br>[2001]<br>        type=endpoint<br>        context=out-local<br>        disallow=all<br>        allow=ulaw<br>        allow=alaw<br>        transport=system-udp<br>        auth=2001<br>        aors=2001<br>        direct_media=no<br>        rtp_symmetric=yes<br>        force_rport=yes<br>        allow=alaw<br>        allow=speex<br>        allow=speex16<br>        allow=speex32<br>        allow=gsm<br><br><br>[2001]<br>        type=aor<br>        qualify_frequency=5000<br>        authenticate_qualify=yes<br>        max_contacts=1<br>        remove_existing=yes<br><br>[2001]<br>        type=auth<br>        auth_type=userpass<br>        password=test<br>        username=test<br><br></div>Best Regards,<br></div>Madushan</div>
</blockquote></div><br></div>