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>

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