[Freeswitch-users] Inbound call one way audio, outbound call works
Paul Muaddib
paul.muaddib83 at gmail.com
Wed Jul 4 22:10:01 UTC 2018
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 at 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
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