<div dir="ltr"><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">Hi,</div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">On a security side note, don't forget to replace your, and your peer's, real IP address before posting on public mailling lists.</div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">About your issue, Can you tell that INVITE packet size ?</div><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">Try to grep it out of a tcpdump:</div><div class="gmail_default" style=""><font face="tahoma, sans-serif">"tcpdump port 5060 -nnn | grep YOUR_PEER_IP_HERE"</font><br></div><div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_default" style="font-family:tahoma,sans-serif;font-size:small">(apt-get install tcpdump, if you need)</div><br clear="all"><div><div class="gmail_signature"><div dir="ltr"><div><font face="tahoma, sans-serif"><br></font></div><div><font face="tahoma, sans-serif" style="background-color:rgb(255,255,255)">---</font></div><div><font face="tahoma, sans-serif" style="background-color:rgb(255,255,255)">Marco<br></font></div><div><span style="background-color:rgb(255,255,255)"><font face="tahoma, sans-serif"><div>---</div><br></font></span></div></div></div></div>
<br><div class="gmail_quote">On Sat, Jan 2, 2016 at 7:56 AM, <span dir="ltr"><<a href="mailto:j.feugne@prisnets.ch" target="_blank">j.feugne@prisnets.ch</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">hello,<br>
I'm just starting with Sipwise and I've setup a VM running version mr4.1.1 CE.<br>
I have followed the instructions of the manual and all subscribers can call each others.<br>
I also setup a sip peer to voicetrading, and here is waht I have withthe command line ngrep sip on my vm while placing and outband call:<br>
NB, voicetrading is expecting for premium rate all number to be leading with 00000 (5 zero)<br>
<br>
U <a href="http://84.253.60.150:5060" rel="noreferrer" target="_blank">84.253.60.150:5060</a> -> <a href="http://77.72.169.129:5060" rel="noreferrer" target="_blank">77.72.169.129:5060</a><br>
INVITE sip:0000041227312221@77.72.169.129:5060;transport=udp SIP/2.0..Record-Route: <sip:84.253.60.150;r2=on;<br>
lr=on;ftag=2AF8D981-568781960002DB55-51003700;ngcplb=yes>..Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=2AF8<br>
D981-568781960002DB55-51003700;ngcplb=yes>..Via: SIP/2.0/UDP 84.253.60.150;branch=z9hG4bK603f.7793d5b01aaccef<br>
36467f073399c00c2.0..Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKVg2PkaeG;rport=5080..Fr<br>
om: <<a href="mailto:sip%3A41225017070@prisnets.adoovoice.com" target="_blank">sip:41225017070@prisnets.adoovoice.com</a>>;tag=2AF8D981-568781960002DB55-51003700..To: <sip:000004122731222<br>
<a href="mailto:1@77.72.169.129" target="_blank">1@77.72.169.129</a>>..CSeq: 11 INVITE..Call-ID: scRGTP71RF5cdeju0O0qmw.._b2b-1..Max-Forwards: 69..Allow: INVITE,<br>
ACK, CANCEL, BYE, NOTIFY, MESSAGE, OPTIONS, SUBSCRIBE..Supported: replaces, norefersub, extended-refer, outbo<br>
und, path, X-cisco-serviceuri..P-Asserted-Identity: <<a href="mailto:sip%3A41225017070@prisnets.adoovoice.com" target="_blank">sip:41225017070@prisnets.adoovoice.com</a>>..Authorization:<br>
Digest username="", realm="<a href="http://prisnets.adoovoice.com" rel="noreferrer" target="_blank">prisnets.adoovoice.com</a>", nonce="4087963395", uri="<a href="mailto:sip%3A0000041227312221@77.72.169.1" target="_blank">sip:0000041227312221@77.72.169.1</a><br>
29:5060;transport=udp", response="c2fb4cd3a1235efd35e99190aedc45c2", algorithm=MD5..Content-Type: application<br>
/sdp..Content-Length: 322..Contact: <<a href="mailto:sip%3Angcp-lb@84.253.60.150" target="_blank">sip:ngcp-lb@84.253.60.150</a>:5060;ngcpct=7369703a3132372e302e302e313a353038<br>
30>....v=0..o=Zoiper 0 0 IN IP4 84.253.60.150..s=Zoiper..c=IN IP4 84.253.60.150..t=0 0..m=audio 32468 RTP/AVP<br>
3 0 8 18 101..a=rtpmap:3 GSM/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp<br>
:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=sendrecv..a=rtcp:32469..a=direction:both<br>
..<br>
<br>
<br>
After almost 1 minute with no ringing sound nor answer the call hang up.<br>
<br>
Can someone please help me point out where the problem is comming and how I can solve it?<br>
<br>
Thank you<br>
<br>
Jephthé Feugné<br>
_______________________________________________<br>
Spce-user mailing list<br>
<a href="mailto:Spce-user@lists.sipwise.com" target="_blank">Spce-user@lists.sipwise.com</a><br>
<a href="https://lists.sipwise.com/listinfo/spce-user" rel="noreferrer" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br>
</blockquote></div><br></div></div>