I've just posted the output of the call from fs_cli, you can have a look on patebin.freeswitch.org. The post is from fransjos. Thank you NDUWAYEZU Joselyne 2016-04-29 22:15 GMT+02:00 Michael Collins <msc@freeswitch.org>:
Hello,
That is a SIP trace, which in this case isn't going to help you. You need to see the output from fs_cli or /var/log/freeswitch.log on a failed call. Capture the output from your terminal program or copy out of freeswitch.log. Copy that into pastebin.freeswitch.org. Try username of "pastebin" and password "freeswitch"
-MSC
On Fri, Apr 29, 2016 at 6:29 AM, Nduwayezu, Joselyne <35633@heb.be> wrote:
Hello, I can not log on pastebin.freeswitch.org This is the log when i make a call
I still having the problem
root@back-1:/usr/local/freeswitch/conf/dialplan/public# ngrep -d eth0 -t -W byline "$1" port 5060 -q interface: eth0 (10.0.0.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) U 2016/04/29 13:14:47.605641 10.0.0.5:5060 -> 10.0.0.6:5080 OPTIONS sip:10.0.0.6:5080 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0. To: sip:10.0.0.6:5080. From: <sip:opensips@10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e. CSeq: 10 OPTIONS. Call-ID: 6d19857b67133c70-6120@10.0.0.5. Max-Forwards: 70. Content-Length: 0. User-Agent: OpenSIPS (2.1.2 (x86_64/linux)). .
U 2016/04/29 13:14:47.606003 10.0.0.6:5080 -> 10.0.0.5:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKf8a4.0f4ccc25.0. From: <sip:opensips@10.0.0.5>;tag=5f71290c3118e5f12e8e1e2bda5b046a-b78e. To: <sip:10.0.0.6:5080>;tag=accp7Bpp1m4vN. Call-ID: 6d19857b67133c70-6120@10.0.0.5. CSeq: 10 OPTIONS. Contact: <sip:10.0.0.6:5080>. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. .
U 2016/04/29 13:14:50.064942 10.0.0.4:5060 -> 10.0.0.6:5080 INVITE sip:opensips@10.0.0.4:5060;transport=udp SIP/2.0. Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>. Call-ID: 03545-MR-1fc39fe6-23cc798d3@sip3.ovh.fr. Contact: <sip:91.121.129.159:5060>. Content-Type: application/sdp. CSeq: 493433999 INVITE. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06. Max-Forwards: 26. Record-Route: <sip:91.121.129.159:5060;lr>. To: <sip:0972550428@10.7.1.68;user=phone>. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd. Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK. User-Agent: Cirpack/v4.70 (gw_sip). Content-Length: 445. . v=0. o=cp10 146193569025 146193569025 IN IP4 10.7.1.122. s=SIP Call. c=IN IP4 91.121.129.155. t=0 0. m=audio 30178 RTP/AVP 18 4 0 8 125 111 101. b=AS:21. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000/1. a=fmtp:4 annexa=no. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:125 CLEARMODE/8000/1. a=rtpmap:111 iLBC/8000/1. a=fmtp:111 mode=30. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv.
U 2016/04/29 13:14:50.065335 10.0.0.6:5080 -> 10.0.0.4:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd. Record-Route: <sip:10.0.0.4;lr;did=ba7.1907a81>. Record-Route: <sip:91.121.129.159:5060;lr>. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06. To: <sip:0972550428@10.7.1.68;user=phone>. Call-ID: 03545-MR-1fc39fe6-23cc798d3@sip3.ovh.fr. CSeq: 493433999 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Content-Length: 0. .
U 2016/04/29 13:14:50.067423 10.0.0.6:5080 -> 10.0.0.4:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-GPEV-34d2c7da-22bfa7cd. Max-Forwards: 26. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06. To: <sip:0972550428@10.7.1.68;user=phone>;tag=BN5e966SyXtFH. Call-ID: 03545-MR-1fc39fe6-23cc798d3@sip3.ovh.fr. CSeq: 493433999 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "opensips" <sip:opensips@10.7.1.68
;party=calling;privacy=off;screen=no. .
U 2016/04/29 13:14:50.068695 10.0.0.4:5060 -> 10.0.0.6:5080 ACK sip:opensips@10.0.0.4:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK23f3.5dd366.0. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=03545-JU-1fc39fe7-19ee23e06. Call-ID: 03545-MR-1fc39fe6-23cc798d3@sip3.ovh.fr. To: <sip:0972550428@10.7.1.68;user=phone>;tag=BN5e966SyXtFH. CSeq: 493433999 ACK. Max-Forwards: 70. User-Agent: OpenSIPS (2.1.2 (x86_64/linux)). Content-Length: 0. .
U 2016/04/29 13:14:52.030194 10.0.0.4:5060 -> 10.0.0.6:5080 INVITE sip:opensips@10.0.0.4:5060;transport=udp SIP/2.0. Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>. Call-ID: 13530-HA-1fc3a093-020bb53b6@sip3.ovh.fr. Contact: <sip:91.121.129.159:5060>. Content-Type: application/sdp. CSeq: 493434154 INVITE. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46. Max-Forwards: 26. Record-Route: <sip:91.121.129.159:5060;lr>. To: <sip:0972550428@10.7.1.68;user=phone>. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-FZAL-34d2c970-111a1872. Allow: REFER,INVITE,NOTIFY,ACK,UPDATE,OPTIONS,REGISTER,SUBSCRIBE,NOTIFY,CANCEL,BYE,PRACK. User-Agent: Cirpack/v4.70 (gw_sip). Content-Length: 315. . v=0. o=cp10 146193569231 146193569231 IN IP4 10.7.16.156. s=SIP Call. c=IN IP4 91.121.129.144. t=0 0. m=audio 33498 RTP/AVP 18 0 8 101. b=AS:21. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000/1. a=rtpmap:8 PCMA/8000/1. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv.
U 2016/04/29 13:14:52.030438 10.0.0.6:5080 -> 10.0.0.4:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-FZAL-34d2c970-111a1872. Record-Route: <sip:10.0.0.4;lr;did=8df.224b90a>. Record-Route: <sip:91.121.129.159:5060;lr>. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46. To: <sip:0972550428@10.7.1.68;user=phone>. Call-ID: 13530-HA-1fc3a093-020bb53b6@sip3.ovh.fr. CSeq: 493434154 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Content-Length: 0. .
U 2016/04/29 13:14:52.032086 10.0.0.6:5080 -> 10.0.0.4:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0. Via: SIP/2.0/UDP 91.121.129.159:5060 ;branch=z9hG4bK-FZAL-34d2c970-111a1872. Max-Forwards: 26. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46. To: <sip:0972550428@10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c. Call-ID: 13530-HA-1fc3a093-020bb53b6@sip3.ovh.fr. CSeq: 493434154 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. Remote-Party-ID: "opensips" <sip:opensips@10.7.1.68
;party=calling;privacy=off;screen=no. .
U 2016/04/29 13:14:52.032599 10.0.0.4:5060 -> 10.0.0.6:5080 ACK sip:opensips@10.0.0.4:5060;transport=udp SIP/2.0. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK0efa.a89ffd15.0. From: "003228800555" <sip:003228800555@sip3.ovh.fr ;user=phone>;tag=13530-NO-1fc3a094-6e7e7dc46. Call-ID: 13530-HA-1fc3a093-020bb53b6@sip3.ovh.fr. To: <sip:0972550428@10.7.1.68;user=phone>;tag=cyy7a2QXU6g2c. CSeq: 493434154 ACK. Max-Forwards: 70. User-Agent: OpenSIPS (2.1.2 (x86_64/linux)). Content-Length: 0. .
U 2016/04/29 13:15:00.670131 10.0.0.4:5060 -> 10.0.0.6:5080 OPTIONS sip:10.0.0.6:5080 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0. To: sip:10.0.0.6:5080. From: <sip:opensips@10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59. CSeq: 10 OPTIONS. Call-ID: 24a5dabd71563ec7-8608@10.0.0.4. Max-Forwards: 70. Content-Length: 0. User-Agent: OpenSIPS (2.1.2 (x86_64/linux)). .
U 2016/04/29 13:15:00.670514 10.0.0.6:5080 -> 10.0.0.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.0.0.4:5060;branch=z9hG4bK03bd.4fd62c01.0. From: <sip:opensips@10.0.0.4>;tag=ea1fc35981b38c23d4118e13e0c2a171-fd59. To: <sip:10.0.0.6:5080>;tag=D7Q0cX80rF7mr. Call-ID: 24a5dabd71563ec7-8608@10.0.0.4. CSeq: 10 OPTIONS. Contact: <sip:10.0.0.6:5080>. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. .
U 2016/04/29 13:15:09.009870 10.0.0.6:5080 -> 10.0.0.5:5060 OPTIONS sip:10.0.0.5;transport=udp SIP/2.0. Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK2vXj0Nc68SXXQ. Max-Forwards: 70. From: <sip:10.0.0.5>;tag=egHSerS4NrX7K. To: <sip:10.0.0.5>. Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0. CSeq: 90631748 OPTIONS. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. .
U 2016/04/29 13:15:09.009969 10.0.0.6:5080 -> 10.0.0.4:5060 OPTIONS sip:10.0.0.4;transport=udp SIP/2.0. Via: SIP/2.0/UDP 10.0.0.6:5080;rport;branch=z9hG4bK35pB2gX952KgK. Max-Forwards: 70. From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF. To: <sip:10.0.0.4>. Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0. CSeq: 90631749 OPTIONS. User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20160404T210025Z~f32edbb936~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. .
U 2016/04/29 13:15:09.010629 10.0.0.4:5060 -> 10.0.0.6:5080 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.0.0.6:5080 ;received=10.0.0.6;rport=5080;branch=z9hG4bK35pB2gX952KgK. From: <sip:10.0.0.4>;tag=FSajgKa8j1KtF. To: <sip:10.0.0.4>;tag=61890dad1e908c702027bf054a266115.990f. Call-ID: 3d12b080-88af-1234-b492-000d3a2233e0. CSeq: 90631749 OPTIONS. Server: OpenSIPS (2.1.2 (x86_64/linux)). Content-Length: 0. .
U 2016/04/29 13:15:09.011254 10.0.0.5:5060 -> 10.0.0.6:5080 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.0.0.6:5080 ;received=10.0.0.6;rport=5080;branch=z9hG4bK2vXj0Nc68SXXQ. From: <sip:10.0.0.5>;tag=egHSerS4NrX7K. To: <sip:10.0.0.5>;tag=2bf6dd58f2c27bac032ef66b671d14ff.52bb. Call-ID: 3d12aa3e-88af-1234-b492-000d3a2233e0. CSeq: 90631748 OPTIONS. Server: OpenSIPS (2.1.2 (x86_64/linux)). Content-Length: 0. .
NDUWAYEZU Joselyne
2016-04-29 0:58 GMT+02:00 Michael Collins <msc@freeswitch.org>:
It sounds like the dialplan never makes it to your extension. Get a debug log of the call and put it on pastebin.freeswitch.org. We can probably help you figure it out from there.
-MSC
On Thu, Apr 28, 2016 at 12:40 AM, Nduwayezu, Joselyne <35633@heb.be> wrote:
Hello,
I would like to invoke php script in the dial plan. I defined the dialplan as follos:
<include> <extension name="outbound-socket"> <condition field="${sip_to_user}" expression="(972550428)$"> <action application="set" data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/> <action application="socket" data="10.0.0.6:8021 async full"/>--> </condition> </extension> </include>
And my php script is:
<?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: playback\n"; echo "execute-app-arg: \ /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n \ "; // Wait sleep(5); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?>
When i dial the DID number, the call state is "call setup" for a while, and after i'm asked to leave a message because the number is busy or unreachable.
I tried different sockets in the dialplan (5060, 5080, 8040) with no results
Any adeas of what is happening?
Thanks
NDUWAYEZU Joselyne
Haute École de Bruxelles
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Haute École de Bruxelles _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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