Hi Kaleem,

thank you for your reply. I solved my problem with miniupnpd. I actually hoped to do it without any firewall rules. I thought when I register SIP with my gateway provider that freeswitch would initiate the audio via rtp over UDP and I don’t have not to change anything. But my provider always starts the audio. So it seems to be there is no way around some firewall rules even though it is just during the call. Whats nice though is that I only open ports for RTP during the call by limiting the port range for miniupnpd. SIP is keep open by freeswitch without any forwarding. Odd for me is that the RTCP data is not accepted from my provider. Freeswitch is not opening any port via upnpd for it.

Regards,
Paul


2018-07-10 15:58 GMT+02:00 kaleem rehman <k4kaleem@gmail.com>:
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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