Hi Paul, its likely to be firewall blocking ports. looking at the log entry { *m=audio 19394 RTP/AVP 8 100* } other party is using RTP port : 19394 , looking at port, its likely they are using RTP range of 16384 to 32767, is this port allowed on your side? as this is the standard RTP range. regards, Kaleem ---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Wed, 4 Jul 2018 18:48:50 +0200 Subject: Re: [Freeswitch-users] 481 Call does not Exit Solved it. 2018-07-03 22:27 GMT+02:00 Paul Muaddib <paul.muaddib83@gmail.com>:
Hi,
if I call someone, who accepts the call and then hangs up, the call does not end on my side
Error message: 481 Call does not Exit
What is the reason for this?
Best regards, Paul
---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 00:10:01 +0200 Subject: [Freeswitch-users] Inbound call one way audio, outbound call works Hi, when I get inbound calls I can here the caller but the caller can't here the callee Other way around, making outbound calls is not a problem. Caller and callee can hear each other (My Setup) Phone -> FS -> NAT -> Gateway <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> What's the problem? Name Type Data State ================================================================================================= 10.0.200.2 alias internal ALIASED external profile sip:mod_sofia@10.0.200.2:5080 RUNNING (0) external::sip-trunk.telekom.de gateway sip:XXXXXXXXXXXX@sip-trunk.telekom.de REGED internal profile sip:mod_sofia@10.0.200.2:5060 RUNNING (0) ================================================================================================= sofia status profile internal ================================================================================================= Name internal Domain Name N/A Auto-NAT true DBName sofia_reg_internal Pres Hosts 10.0.200.2,10.0.200.2 Dialplan XML Context public Challenge Realm auto_from RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5060 BIND-URL sip:mod_sofia@10.0.200.2:5060;transport=udp,tcp WS-BIND-URL sip:mod_sofia@10.0.200.2:5066;transport=ws WSS-BIND-URL sips:mod_sofia@10.0.200.2:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE info CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 6 FAILED-CALLS-IN 1 CALLS-OUT 6 FAILED-CALLS-OUT 4 REGISTRATIONS 7 sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT true DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 10.0.200.2 Ext-RTP-IP 87.157.X.X SIP-IP 10.0.200.2 Ext-SIP-IP 87.157.X.X URL sip:mod_sofia@10.0.200.2:5080 BIND-URL sip:mod_sofia@10.0.200.2:5080;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 3 FAILED-CALLS-IN 1 CALLS-OUT 2 FAILED-CALLS-OUT 0 REGISTRATIONS 0 nat_map status Nat Type: NAT-PMP, ExtIP: 87.157.68.118 NAT port mapping enabled. I only map RTP ports. SIP is registered via TCP Console debug level 7 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7091 Duplicate SDP v=0 o=- 0 2 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4767 sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de. 2018-07-04 23:12:24.905378 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:24.925380 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:24.945382 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [ready][200] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [received][100] 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7094 Remote SDP: v=0 o=- 0 3 IN IP4 217.0.15.67 s=on transit c=IN IP4 217.0.132.134 t=0 0 m=audio 19394 RTP/AVP 8 100 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=maxptime:20 a=ptime:20 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4365 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4708 Set telephone-event payload to 100@8000 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4767 sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de Set 2833 dtmf send payload to 100 recv payload to 100 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/external/+49XXXXXXXXXX@sip-trunk.telekom.de. 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7999 Processing updated SDP 2018-07-04 23:12:31.625959 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [completed][200] 2018-07-04 23:12:31.665962 [DEBUG] sofia.c:7084 Channel sofia/external/+ 49XXXXXXXXXX@sip-trunk.telekom.de entering state [ready][200] I am trying for days now but google is no real help. Regards, Paul ---------- Forwarded message ---------- From: Paul Muaddib <paul.muaddib83@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Cc: Bcc: Date: Thu, 5 Jul 2018 14:53:46 +0200 Subject: [Freeswitch-users] LOSE_RACE, 3436 Originate Failed. Cause: NONE Hi, sometimes I have the rare case that I lose a call. I can't reproduce it. It seems a bit random. When this happens all group members have a LOSE_RACE and the caller gets disconnected? Why? Thanks for helping Regards, Paul <extension name="Operator"> <condition field="destination_number" expression="^(operator)$|^(90)$" require-nested="false"> <action application="export" data="dialed_extension=operator"/> <action application="set" data="dialed_user=$1@${domain_name}"/> <action application="set" data="call_timeout=90"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bind_digit_action" data="get_digits,~^([1-9][0-9])$,exec:lua,get_digits.lua,peer,self"/> <action application="digit_action_set_realm" data="get_digits"/> <action application="ring_ready"/> <condition field="${office_status}" expression="^(open)$" break="on-true"> <action application="bridge" data="group/buero :_: pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition> <condition field="${office_status}" expression="^(closed)$" break="on-true"> <action application="bridge" data="group/buero :_: group/werkstatt,pickup/global"/> <action application="hangup" data="NO_ANSWER"/> </condition> </condition> </extension> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org