<div dir="ltr">Thanks a bunch Ken !<div><br></div><div>I will try that tomorrow to see what exactly is happening.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Feb 18, 2015 at 9:22 PM, Ken Rice <span dir="ltr"><<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<font face="Monaco, Courier New"><span style="font-size:11pt">That actually looks like an error from the PSTN side... 603 is global failure... And means don’t even bother trying this call again...<br>
<br>
The real way to figure this out from FreeSWITCH is get on fs_cli and enable global sip tracing in sofia so you can see whats going on.<span class=""><br>
<br>
<br>
On 2/18/15, 9:08 AM, "huseyin kalyoncu" <<a href="http://hkalyoncu@gmail.com" target="_blank">hkalyoncu@gmail.com</a>> wrote:<br>
<br>
</span></span></font><blockquote><span class=""><font face="Monaco, Courier New"><span style="font-size:11pt">what about the PSTN side? did you check that? <br>
did call ever pass to the PSTN side? might be the error came from PSTN side?<br>
<br>
On Wed, Feb 18, 2015 at 4:28 PM, Rahul MathuR <<a href="http://rahul.ultimate@gmail.com" target="_blank">rahul.ultimate@gmail.com</a>> wrote:<br>
</span></font></span><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt"><span class="">Hi all,<br>
<br>
I was doing a POC of WebRTC based audio call to PSTN, routed via kamailio (for protocol translation & proxy) and FS as SIP server.<br>
I get 100 trying from FS but after that 603 decline with reason - "Reason: Q.850;cause=16;text="NORMAL_CLEARING""<br>
<br>
Also, my phone didn't ring whereas, ITU T Q850 codes say,<br>
16 NORMAL_CLEARING <br>
normal call clearing [Q.850] This cause indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this cause is not the network.<br>
<br>
<br>
Could you please help me in resolving this issue.<br>
<br>
<br>
<br>
The messages are as under - <br>
<br>
1. INVITE from JsSIP to Kamailio<br>
<br></span>
INVITE sip:00919650926333 <tel:00919650926333> @<FreeSwitch_IP> SIP/2.0<span class=""><br>
Route: <sip:<Kamailio_IP>:10080;transport=ws;lr><br>
Via: SIP/2.0/TCP amadf8lur89p.invalid;branch=z9hG4bK1158107<br>
Max-Forwards: 69<br></span>
To: <sip:00919650926333 <tel:00919650926333> @<FreeSwitch_IP>><span class=""><br>
From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=3dp7hdgg6j<br>
Call-ID: n41t8s01dnclcbodd3il<br>
CSeq: 352 INVITE<br></span>
Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="d0593b9e-b772-11e4-aeec-2965abc3007e", uri="sip:00919650926333 <tel:00919650926333> @<FreeSwitch_IP>", response="12e4164f793098007d8bf09f7b50815f", qop=auth, cnonce="l02s3ec50q1g", nc=00000001<span class=""><br>
Contact: <sip:<a href="http://220bscel@amadf8lur89p.invalid" target="_blank">220bscel@amadf8lur89p.invalid</a>;transport=ws;ob><br>
Content-Type: application/sdp<br>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br>
Supported: ice,outbound<br>
User-Agent: JsSIP 0.6.4<br>
Content-Length: 1769<br>
<br>
v=0<br>
o=- 9056480915531460217 2 IN IP4 127.0.0.1<br>
s=-<br>
t=0 0<br>
a=group:BUNDLE audio<br>
a=msid-semantic: WMS dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG<br>
m=audio 13228 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126<br>
c=IN IP4 59.178.158.4<br>
a=rtcp:13228 IN IP4 59.178.158.4<br></span>
a=candidate:2437072876 1 udp <a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a> <tel:<a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a>> 192.168.1.2 64540 typ host generation 0<br>
a=candidate:2437072876 2 udp <a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a> <tel:<a href="tel:2122260223" value="+12122260223" target="_blank">2122260223</a>> 192.168.1.2 64540 typ host generation 0<div><div class="h5"><br>
a=candidate:3753982748 1 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0<br>
a=candidate:3753982748 2 tcp 1518280447 192.168.1.2 0 typ host tcptype active generation 0<br>
a=candidate:941443129 1 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0<br>
a=candidate:941443129 2 udp 1686052607 59.178.158.4 13228 typ srflx raddr 192.168.1.2 rport 64540 generation 0<br>
a=ice-ufrag:9Yhi1W5j+XrHKQKQ<br>
a=ice-pwd:fVw9fkXXv1bteXnWh3B/694c<br>
a=ice-options:google-ice<br>
a=fingerprint:sha-256 C1:96:B0:69:7A:4C:D6:3B:DD:6C:4B:83:BF:F6:45:56:43:95:B4:46:E0:11:BF:AB:2A:42:8D:47:F7:DA:A8:66<br>
a=setup:actpass<br>
a=mid:audio<br>
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level<br>
a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time" target="_blank">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a><br>
a=sendrecv<br>
a=rtcp-mux<br>
a=rtpmap:111 opus/48000/2<br>
a=fmtp:111 minptime=10<br>
a=rtpmap:103 ISAC/16000<br>
a=rtpmap:104 ISAC/32000<br>
a=rtpmap:9 G722/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:106 CN/32000<br>
a=rtpmap:105 CN/16000<br>
a=rtpmap:13 CN/8000<br>
a=rtpmap:126 telephone-event/8000<br>
a=maxptime:60<br>
a=ssrc:1011570965 cname:m/wi9qp3sxYVEQLV<br>
a=ssrc:1011570965 msid:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG 9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4<br>
a=ssrc:1011570965 mslabel:dcrNk6emj9gTfynUdLaYmsTVbZwnKc9iEwCG<br>
a=ssrc:1011570965 label:9fd4df2c-5c5d-4dec-a1a6-e181d5da35c4<br>
<br>
<br>
2. INVITE from Kamailio to FS<br>
<br></div></div>
INVITE sip:00919560509733 <tel:00919560509733> @<FreeSwitch_IP> SIP/2.0<span class=""><br>
Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on><br>
Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on><br>
Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0<br>
Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437<br>
Max-Forwards: 68<br></span>
To: <sip:00919560509733 <tel:00919560509733> @<FreeSwitch_IP>><span class=""><br>
From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej<br>
Call-ID: pk2gvvb0qancp1ki8v5t<br>
CSeq: 5489 INVITE<br></span>
Proxy-Authorization: Digest algorithm=MD5, username="55555", realm="<FreeSwitch_IP>", nonce="85496252-b74d-11e4-ae0e-2965abc3007e", uri="sip:00919560509733 <tel:00919560509733> @<FreeSwitch_IP>", response="a821425e292b05c4b9480cb66336cb93", qop=auth, cnonce="h8i58dtuuc0p", nc=00000001<div><div class="h5"><br>
Contact: <sip:<a href="http://g5uvomah@4dcddrn8nrh8.invalid" target="_blank">g5uvomah@4dcddrn8nrh8.invalid</a>;transport=ws;ob;alias=59.178.154.62~12016~5><br>
Content-Type: application/sdp<br>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS<br>
Supported: ice,outbound<br>
User-Agent: JsSIP 0.6.4<br>
Content-Length: 782<br>
<br>
v=0<br>
o=- 4750671857217657811 2 IN IP4 <Kamailio_IP><br>
s=-<br>
t=0 0<br>
a=msid-semantic: WMS IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te<br>
m=audio 32838 RTP/AVP 111 103 104 9 0 8 106 105 13 126<br>
c=IN IP4 <Kamailio_IP><br>
a=rtpmap:111 opus/48000/2<br>
a=fmtp:111 minptime=10<br>
a=rtpmap:103 ISAC/16000<br>
a=rtpmap:104 ISAC/32000<br>
a=rtpmap:9 G722/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:106 CN/32000<br>
a=rtpmap:105 CN/16000<br>
a=rtpmap:13 CN/8000<br>
a=rtpmap:126 telephone-event/8000<br>
a=maxptime:60<br></div></div>
a=ssrc:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a> <tel:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a>> cname:Bk0e7lSPEkClAxNy<br>
a=ssrc:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a> <tel:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a>> msid:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te 291b0246-4d75-4347-8b2d-e8e42171a36e<br>
a=ssrc:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a> <tel:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a>> mslabel:IMbls497PsJs5HOf2sY8de0vWcfW3Vjim2Te<br>
a=ssrc:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a> <tel:<a href="tel:2268746120" value="+12268746120" target="_blank">2268746120</a>> label:291b0246-4d75-4347-8b2d-e8e42171a36e<div><div class="h5"><br>
a=sendrecv<br>
a=rtcp:32839<br>
<br>
<br>
3. 100 trying from FS to Kamailio<br>
<br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0<br>
Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437<br>
Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:5090;r2=on;lr=on><br>
Record-Route: <sip:A4c7QkcuqY2hdQV9Y7p+J2A7spo+LvA=@<Kamailio_IP>:10080;transport=ws;r2=on;lr=on><br>
From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej<br>
To: <sip:00919560509733@<FreeSwitch_IP>><br>
Call-ID: pk2gvvb0qancp1ki8v5t<br>
CSeq: 5489 INVITE<br>
User-Agent: ASTPP<br>
Content-Length: 0<br>
<br>
<br>
4. 603 from FS to Kamailio<br>
<br>
SIP/2.0 603 Decline<br>
Via: SIP/2.0/UDP <Kamailio_IP>:5090;branch=z9hG4bKb937.6279f7ec5448a6c7867563e9b50fbe39.0<br>
Via: SIP/2.0/TCP 4dcddrn8nrh8.invalid;rport=12016;received=59.178.154.62;branch=z9hG4bK4195437<br>
Max-Forwards: 68<br>
From: "55555" <sip:55555@<FreeSwitch_IP>>;tag=i5so9j18ej<br>
To: <sip:00919560509733@<FreeSwitch_IP>>;tag=y4D44m427U2Kj<br>
Call-ID: pk2gvvb0qancp1ki8v5t<br>
CSeq: 5489 INVITE<br>
User-Agent: ASTPP<br>
Accept: application/sdp<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, NOTIFY<br>
Supported: path, replaces<br>
Allow-Events: talk, hold, conference, refer<br>
Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>
Content-Length: 0<br>
Remote-Party-ID: "00919560509733" <sip:00919560509733@<FreeSwitch_IP>>;party=calling;privacy=off;screen=no<br>
</div></div></span></font></blockquote></blockquote><span class="HOEnZb"><font color="#888888"><font face="Monaco, Courier New"><span style="font-size:11pt"><br>
-- <br>
Ken<br>
<font color="#0000FF"><u><a href="http://www.FreeSWITCH.org" target="_blank">http://www.FreeSWITCH.org</a><br>
<a href="http://www.ClueCon.com" target="_blank">http://www.ClueCon.com</a><br>
<a href="http://www.OSTAG.org" target="_blank">http://www.OSTAG.org</a><br>
</u></font><a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br>
Twitter: @FreeSWITCH<br>
<br>
</span></font>
</font></span></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature">Warm Regds.<br>MathuRahul<br></div>
</div>