[Freeswitch-users] 481 Call does not Exit
Paul Muaddib
paul.muaddib83 at gmail.com
Tue Jul 10 17:11:06 UTC 2018
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 at 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 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at 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 at 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 at 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 at 10.0.200.2:5060
> BIND-URL sip:mod_sofia at 10.0.200.2:5060;transport=udp,tcp
> WS-BIND-URL sip:mod_sofia at 10.0.200.2:5066;transport=ws
> WSS-BIND-URL sips:mod_sofia at 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 at 10.0.200.2:5080
> BIND-URL sip:mod_sofia at 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 at 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 at 8000
> 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4708 Set
> telephone-event payload to 100 at 8000
> 2018-07-04 23:12:24.905378 [DEBUG] switch_core_media.c:4767
> sofia/external/+49XXXXXXXXXX at 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 at 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 at sip-trunk.telekom.de entering state [completed][200]
> 2018-07-04 23:12:24.945382 [DEBUG] sofia.c:7084 Channel sofia/external/+
> 49XXXXXXXXXX at sip-trunk.telekom.de entering state [ready][200]
> 2018-07-04 23:12:31.605957 [DEBUG] sofia.c:7084 Channel sofia/external/+
> 49XXXXXXXXXX at 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 at 8000
> 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4708 Set
> telephone-event payload to 100 at 8000
> 2018-07-04 23:12:31.605957 [DEBUG] switch_core_media.c:4767
> sofia/external/+49XXXXXXXXXX at 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 at 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 at sip-trunk.telekom.de entering state [completed][200]
> 2018-07-04 23:12:31.665962 [DEBUG] sofia.c:7084 Channel sofia/external/+
> 49XXXXXXXXXX at 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 at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at 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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>
>
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