<div dir="ltr">yes I have unchecked it.<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 9, 2016 at 10:27 PM, Administrator TOOTAI <span dir="ltr"><<a href="mailto:admin@tootai.net" target="_blank">admin@tootai.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Le 09/09/2016 à 18:32, Madushan Geethanga a écrit :<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
</blockquote>
<br>
If you're not using RTP encryption did you uncheck the option in your RTP TAB from identity ?<br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span class="">
<br>
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" rel="noreferrer" target="_blank">123.231.72.210:33878</a><br></span>
<<a href="http://123.231.72.210:33878" rel="noreferrer" target="_blank">http://123.231.72.210:33878</a>> ---><br>
INVITE <a href="mailto:sip%3A0@54.206.59.252" target="_blank">sip:0@54.206.59.252</a> <mailto:<a href="mailto:sip%253A0@54.206.59.252" target="_blank">sip%3A0@54.206.59.252</a>><wbr>;user=phone SIP/2.0<span class=""><br>
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9<wbr>hG4bK-bskkkx1t5bas;rport<br>
From: "outburns00-nhvg5vjjn6-2001"<br>
<<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip:outburns00-nhvg5vjjn6-200<wbr>1@54.206.59.252</a><br></span>
<mailto:<a href="mailto:sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip%3Aoutburns00-nhvg5<wbr>vjjn6-2001@54.206.59.252</a>>>;<wbr>tag=1bb809zgaa<br>
To: <<a href="mailto:sip%3A0@54.206.59.252" target="_blank">sip:0@54.206.59.252</a> <mailto:<a href="mailto:sip%253A0@54.206.59.252" target="_blank">sip%3A0@54.206.59.252</a>><wbr>;user=phone><span class=""><br>
Call-ID: 313437333433383639323238313539<wbr>-ahn3begiq66q<br>
CSeq: 1 INVITE<br>
Max-Forwards: 70<br></span>
User-Agent: snom710/<a href="http://8.7.5.35" rel="noreferrer" target="_blank">8.7.5.35</a> <<a href="http://8.7.5.35" rel="noreferrer" target="_blank">http://8.7.5.35</a>><br>
Contact: <<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835" rel="noreferrer" target="_blank">sip:outburns00-nhvg5vjjn6-200<wbr>1@123.231.72.210:45835</a><br>
<<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835" rel="noreferrer" target="_blank">http://sip:outburns00-nhvg5vj<wbr>jn6-2001@123.231.72.210:45835</a>><wbr>>;reg-id=1<div><div class="h5"><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,<br>
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 <a href="tel:2136927789" value="+12136927789" target="_blank">2136927789</a> <a href="tel:2136927789" value="+12136927789" target="_blank">2136927789</a> 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" rel="noreferrer" target="_blank">123.231.72.210:33878</a><br></div></div>
<<a href="http://123.231.72.210:33878" rel="noreferrer" target="_blank">http://123.231.72.210:33878</a>> ---><span class=""><br>
SIP/2.0 401 Unauthorized<br>
Via: SIP/2.0/UDP<br>
123.231.72.210:45835;rport=338<wbr>78;received=123.231.72.210;bra<wbr>nch=z9hG4bK-bskkkx1t5bas<br>
Call-ID: 313437333433383639323238313539<wbr>-ahn3begiq66q<br>
From: "outburns00-nhvg5vjjn6-2001"<br>
<<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip:outburns00-nhvg5vjjn6-200<wbr>1@54.206.59.252</a><br></span>
<mailto:<a href="mailto:sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip%3Aoutburns00-nhvg5<wbr>vjjn6-2001@54.206.59.252</a>>>;<wbr>tag=1bb809zgaa<br>
To: <<a href="mailto:sip%3A0@54.206.59.252" target="_blank">sip:0@54.206.59.252</a><br>
<mailto:<a href="mailto:sip%253A0@54.206.59.252" target="_blank">sip%3A0@54.206.59.252</a>><wbr>;user=phone>;tag=z9hG4bK-bskkk<wbr>x1t5bas<span class=""><br>
CSeq: 1 INVITE<br>
WWW-Authenticate: Digest<br>
realm="asterisk",nonce="147343<wbr>8693/ef923d25464dbedc1dbd85e0c<wbr>cea08b7",opaque="<wbr>210b270d7abb2354",algorithm=<wbr>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" rel="noreferrer" target="_blank">123.231.72.210:33878</a><br></span>
<<a href="http://123.231.72.210:33878" rel="noreferrer" target="_blank">http://123.231.72.210:33878</a>> ---><br>
ACK <a href="mailto:sip%3A0@54.206.59.252" target="_blank">sip:0@54.206.59.252</a> <mailto:<a href="mailto:sip%253A0@54.206.59.252" target="_blank">sip%3A0@54.206.59.252</a>><wbr>;user=phone SIP/2.0<span class=""><br>
Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9<wbr>hG4bK-bskkkx1t5bas;rport<br>
From: "outburns00-nhvg5vjjn6-2001"<br>
<<a href="mailto:sip%3Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip:outburns00-nhvg5vjjn6-200<wbr>1@54.206.59.252</a><br></span>
<mailto:<a href="mailto:sip%253Aoutburns00-nhvg5vjjn6-2001@54.206.59.252" target="_blank">sip%3Aoutburns00-nhvg5<wbr>vjjn6-2001@54.206.59.252</a>>>;<wbr>tag=1bb809zgaa<br>
To: <<a href="mailto:sip%3A0@54.206.59.252" target="_blank">sip:0@54.206.59.252</a><br>
<mailto:<a href="mailto:sip%253A0@54.206.59.252" target="_blank">sip%3A0@54.206.59.252</a>><wbr>;user=phone>;tag=z9hG4bK-bskkk<wbr>x1t5bas<span class=""><br>
Call-ID: 313437333433383639323238313539<wbr>-ahn3begiq66q<br>
CSeq: 1 ACK<br>
Max-Forwards: 70<br></span>
User-Agent: snom710/<a href="http://8.7.5.35" rel="noreferrer" target="_blank">8.7.5.35</a> <<a href="http://8.7.5.35" rel="noreferrer" target="_blank">http://8.7.5.35</a>><br>
Contact: <<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835" rel="noreferrer" target="_blank">sip:outburns00-nhvg5vjjn6-200<wbr>1@123.231.72.210:45835</a><br>
<<a href="http://sip:outburns00-nhvg5vjjn6-2001@123.231.72.210:45835" rel="noreferrer" target="_blank">http://sip:outburns00-nhvg5vj<wbr>jn6-2001@123.231.72.210:45835</a>><wbr>>;reg-id=1<span class=""><br>
Content-Length: 0<br>
<br>
<br>
Best Regards,<br>
Madushan<br>
<br>
<br>
<br>
On Fri, Sep 9, 2016 at 9:53 PM, Madushan Geethanga<br></span><div><div class="h5">
<<a href="mailto:mgliyanage.rc@gmail.com" target="_blank">mgliyanage.rc@gmail.com</a> <mailto:<a href="mailto:mgliyanage.rc@gmail.com" target="_blank">mgliyanage.rc@gmail.co<wbr>m</a>>> wrote:<br>
<br>
Hi,<br>
<br>
I'm trying to setup snom 710 phone with asterisk 13 with PJSIP.<br>
inbound is working fine but i cannot dial out. i don't hear anything<br>
on the phone and asterisk CLI also does not show anything. my config<br>
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>
Best Regards,<br>
Madushan<br>
<br>
<br>
<br>
<br>
</div></div></blockquote>
<br>
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</blockquote></div><br></div>