I changed the following in the ctd.sh script Changed the default of "`printf "v=0\r\no=click-to-dial 0 0 IN IP4 0.0.0.0\r\ns=session\r\nc=IN IP4 0.0.0.0\r\nb=CT:1000\r\nt=0 0\r\nm=audio 9 RTP/AVP 8 0\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:0 PCMU/8000\r\n"`
To "`printf "v=0\r\no=click2dial 0 0 IN IP4 50.XX.XX.156\r\ns=click2dial call\r\nc=IN IP4 173.XX.XX.111\r\nt=0 0\r\nm=audio 12790 RTP/AVP 0 8 18 3 4 97 98\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:18 G729/8000\r\na=rtpmap:97 ilbc/8000\r\na=rtpmap:98 speex/8000\r\n"` And now it is making it into the OpenSIPS/SBC's main route. Not sure why. I noticed another issue now. My snom phone is receiving the INVITE but it is replying with a "404 Not Found" error. (If I test with a Jitsi client I don't have the 404 issue) This shouldn't happen since the TO header is the correct SIP URI. The only thing that can be wrong is that the To: URI is not in <> I think the TM MI function t_uac_dlg isn't placing the <> around the TO: header URI. Reading the RFC I am not 100% sure if the <> are required. U 2012/08/08 22:09:13.756976 192.168.88.1:5060 -> 192.168.88.13:3072 INVITE sip:[email protected]:3072 SIP/2.0. Max-Forwards: 10. Record-Route: <sip:192.168.88.1;r2=on;lr>. Record-Route: <sip:99.XX.XX.161;r2=on;lr>. Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0. Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0. *To: sip:[email protected].* From: <sip:[email protected]>;tag=134448175329440. CSeq: 1 INVITE. Call-ID: 134448175329440.fifouacctd. Content-Length: 226. User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)). Contact: <sip:[email protected]:5060>. Content-Type: application/sdp. . v=0. o=click2dial 0 0 IN IP4 50.XX.XX.156. s=click2dial call. c=IN IP4 173.XX.XX.111. t=0 0. m=audio 12790 RTP/AVP 0 8 18 3 4 97 98. a=rtpmap:0 PCMU/8000. a=rtpmap:18 G729/8000. a=rtpmap:97 ilbc/8000. a=rtpmap:98 speex/8000. # U 2012/08/08 22:09:13.766974 192.168.88.13:3072 -> 192.168.88.1:5060 SIP/2.0 404 Not found. Via: SIP/2.0/UDP 192.168.88.1;branch=z9hG4bK3f03.9cb7ee3.0. Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK3f03.18d165f1.0. From: <sip:[email protected]>;tag=134448175329440. To: <sip:[email protected]>. Call-ID: 134448175329440.fifouacctd. CSeq: 1 INVITE. User-Agent: snom821/8.7.3.10. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, replaces, from-change. Content-Length: 0. On Fri, Jul 27, 2012 at 3:24 PM, <[email protected]> wrote: > Very sure. Normal calls are working with clients behind the OpenSIPS/SBC. > > > > > On , Bogdan-Andrei Iancu <[email protected]> wrote: > > > > > > > > > > > > > > Duane,some stupid question : are you sure your opensips is > > listening on the given IP:port ? have you check with netstat ? > > also have you checked with netstat also if there is traffic queued > > on the sockets ? > > > > > > > > Regards, > > > > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > > > > > On 07/27/2012 12:48 AM, Duane Larson wrote: > > Oh yeah. My first email has the SIPTrace from the > > OpenSIPS/SBC. So I am logged into the OpenSIPS/SBC and did the > > NGREP. So I see it SIP invite (99.XX.XX.161 is the IP of the > > OpenSIPS/SBC). I would even let you log into the OpenSIPS/SBC > > and see it for yourself. Makes no sense. > > > > > > > > U 2012/07/19 18:20:13.486847 50.XX.XX.156:5060 -> 99.XX.XX.161:5060 > > > > INVITE sip:[email protected]:3072;line=g2hfphrk SIP/2.0. > > > > Via: SIP/2.0/UDP 50.57.54.156;branch=z9hG4bKaab1.6c7ffd84.0. > > > > To: sip:[email protected]. > > > > From: sip:[email protected]>;tag=134274001013257. > > > > CSeq: 1 INVITE. > > > > Call-ID: 134274001013257.fifouacctd. > > > > Content-Length: 155. > > > > User-Agent: OpenSIPS (1.8.0-dev0-tls (x86_64/linux)). > > > > Contact: sip:[email protected]:5060>. > > > > Content-Type: application/sdp. > > > > . > > > > v=0. > > > > o=click-to-dial 0 0 IN IP4 0.0.0.0. > > > > s=session. > > > > c=IN IP4 0.0.0.0. > > > > b=CT:1000. > > > > t=0 0. > > > > m=audio 9 RTP/AVP 8 0. > > > > a=rtpmap:8 PCMA/8000. > > > > a=rtpmap:0 PCMU/8000. > > > > > > > > On Thu, Jul 26, 2012 at 4:35 PM, > > Bogdan-Andrei Iancu [email protected]> wrote: > > > > OK, so the OpenSIPS SBC is not receiving > > the INVITE from the OpenSIPS proxy? If so, have you checked on > > the opensips proxy, using network capture tool if the INVITE > > is sent to the proper destination ?? > > > > > > > > > > > > Regards, > > > > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > > > http://www.opensips-solutions.com > > > > > > > > > > > > > > > > > > > > On 07/25/2012 07:58 AM, [email protected] > > wrote: > > > > > > Hey Bogdan, > > > > > > > > I think you might be a little confused from my emails. > > The last email that had a SIP trace and the 100 Trying > > was a click-to-dial generated from php-sip( > http://code.google.com/p/php-sip/) > > and I wanted to show you that with php-sip the OpenSIPS > > server processes the INVITE and replies with a 100 > > Trying. > > > > > > > > Here is what I am doing > > > > > > > > Client_on_LAN OpenSIPS/SBC Internet > > > OpenSIPS/Proxy > > > > > > > > So I am executing the ctd.sh script on the > > OpenSIPS/Proxy server and the INVITE is going to the > > OpenSIPS/SBC server and that is where OpenSIPS isn't > > even seeing the INVITE. > > > > > > > > > > > > > > > > > > > > On , Bogdan-Andrei Iancu [email protected]> > > wrote: > > > > > Hi Duane, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > The INVITE generated by the opensips (triggered by > > the PHP script via MI) will not show up in the opensips > > script - it is directly sent out by opensips internals > > (without script interaction) to the destination from > > DURI / RURI - the only place where you can see it (on > > the opensips instance that generates that INVITE) is by > > using a local route. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > In your capture, I see that there is a 100 trying > > reply received after all from the destination - are > > there any other replies following ? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Bogdan-Andrei Iancu > > > > > > > > > > > > > > > OpenSIPS Founder and Developer > > > > > > > > > > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On 07/23/2012 07:54 PM, [email protected] > > wrote: > > > > > > > > > > > > > > > > > > > > > > > > > That is correct there is no answer to the INVITE > > and it doesn't appear to even enter the main route of > > the OpenSIPS config. I have xlogs set up in the opensips > > script and I never see the INVITE enter. Here is another > > sip trace from the php-sip click to call program and for > > some reason this INVITE does go through the main route > > without issue. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > # > > > > > > > > > > > > > > > U 2012/07/23 11:49:04.398750 50.XX.XX.156:5060 > > -> 99.XX.XX.161:5060 > > > > > > > > > > > > > > > INVITE sip:[email protected]:3072;line=996he62l > > SIP/2.0. > > > > > > > > > > > > > > > Record-Route: > 34533;vst=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA;did=38a.ff1a8d45>. > > > > > > > > > > > > > > > Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0. > > > > > > > > > > > > > > > Via: SIP/2.0/UDP > 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398. > > > > > > > > > > > > > > > From: ;tag=34533. > > > > > > > > > > > > > > > To: sip:[email protected]>. > > > > > > > > > > > > > > > Call-ID: [email protected]. > > > > > > > > > > > > > > > CSeq: 20 INVITE. > > > > > > > > > > > > > > > Contact: . > > > > > > > > > > > > > > > Content-Type: application/sdp. > > > > > > > > > > > > > > > Max-Forwards: 69. > > > > > > > > > > > > > > > User-Agent: PHP SIP. > > > > > > > > > > > > > > > Subject: click2call. > > > > > > > > > > > > > > > Content-Length: 225. > > > > > > > > > > > > > > > P-hint: outbound->inbound . > > > > > > > > > > > > > > > P-hint: Route[6]: mediaproxy . > > > > > > > > > > > > > > > . > > > > > > > > > > > > > > > v=0. > > > > > > > > > > > > > > > o=click2dial 0 0 IN IP4 50.57.75.54. > > > > > > > > > > > > > > > s=click2dial call. > > > > > > > > > > > > > > > c=IN IP4 173.XX.XX.111. > > > > > > > > > > > > > > > t=0 0. > > > > > > > > > > > > > > > m=audio 12790 RTP/AVP 0 8 18 3 4 97 98. > > > > > > > > > > > > > > > a=rtpmap:0 PCMU/8000. > > > > > > > > > > > > > > > a=rtpmap:18 G729/8000. > > > > > > > > > > > > > > > a=rtpmap:97 ilbc/8000. > > > > > > > > > > > > > > > a=rtpmap:98 speex/8000. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > # > > > > > > > > > > > > > > > U 2012/07/23 11:49:04.398750 99.XX.XX.161:5060 > > -> 50.XX.XX.156:5060 > > > > > > > > > > > > > > > SIP/2.0 100 Giving a try. > > > > > > > > > > > > > > > Via: SIP/2.0/UDP 50.XX.XX.156;branch=z9hG4bK10f3.c6fade97.0. > > > > > > > > > > > > > > > Via: SIP/2.0/UDP > 50.XX.XX.54:5065;received=50.XX.XX.54;rport=5065;branch=z9hG4bK185398. > > > > > > > > > > > > > > > From: ;tag=34533. > > > > > > > > > > > > > > > To: sip:[email protected]>. > > > > > > > > > > > > > > > Call-ID: [email protected]. > > > > > > > > > > > > > > > CSeq: 20 INVITE. > > > > > > > > > > > > > > > Server: OpenSIPS (1.8.0-notls (x86_64/linux)). > > > > > > > > > > > > > > > Content-Length: 0. > > > > > > > > > > > > > > > . > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > So I am not sure why the first sip trace INVITE I > > sent isn't being processed but the one above is. Very > > weird. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I haven't tried to send it to another OpenSIPS > > server because I really don't have any other server to > > test with. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On , Bogdan-Andrei Iancu [email protected]> > > wrote: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Hi Duane, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > You mean there is no answer to that INVITE ? > > When you tried to > > > > > > > > > > > > > > > > send the INVITE to another opensips, have you > > noticed errors in > > > > > > > > > > > > > > > > the logs (like parsing errors)? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Bogdan-Andrei Iancu > > > > > > > > > > > > > > > > OpenSIPS Founder and Developer > > > > > > > > > > > > > > > > http://www.opensips-solutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On 07/21/2012 07:45 AM, [email protected] > > wrote: > > > > > > > > > > > > > > > > Has anyone used the ctd.sh example that comes > > with > > > > > > > > > > > > > > > > Opensips in the "example" folder? I am trying > > to use it and the > > > > > > > > > > > > > > > > INVITE gets sent out but nothing happens. I > > even tried with > > > > > > > > > > > > > > > > sending the INVITE to an OpenSIPS server and > > the OpenSIPS server > > > > > > > > > > > > > > > > never even sees it enter the main route even > > though I see that the > > > > > > > > > > > > > > > > INVITE is making it to the server because an > > NGREP shows it making > > > > > > > > > > > > > > > > it. It doesn't make much sense. I even did a > > debug and don't see > > > > > > > > > > > > > > > > anything showing that OpenSIPS sees the > > INVITE. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > <br / -- -- *--*--*--*--*--* Duane *--*--*--*--*--* --
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