I'm troubleshooting Tone detection in Freeswitch installed in AWS (Debian)
I have 2 servers one works the other doesn't:
FreeSWITCH (Version 1.7.0 -928-db61f4d 64bit) DigitalOcean WORKS
FreeSWITCH (Version 1.6.9 -16-d574870 64bit) is ready AWS FAILS
Troubleshooting
Seems to be that Freeswitch in AWS EC2 because of NAT is not able to get the RTP from remote side. When I do a packet capture I don't get any RTP from remote end and in initial SIP SDP from Freeswitch I do see local IP address from AWS even though ext-X params are configured.
1. One instance is installed in AWS, the other one in DigitalOcean,
2. I'm originating my calls using ESL and my Gateway:
originate {ignore_early_media=true}sofia/gateway/asterisk/+14171111111 handle_calls
3. I already configured ext-rtp-ip and ext-sip-ip fields in external.xml
<param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
<param name="ext-sip-ip" value="$${external_sip_ip}"/>
https://www.twilio.com/resources/images/docs/Twilio-Freeswitch.pdf
Any ideas?
Logs
sofia status gateway asterisk
=================================================================================================
Name asterisk
Profile external
Scheme Digest
Realm 192.241.203.21:5090
Username FreeSWITCH
Password no
From <sip:FreeSWITCH@192.241.203.21:5090>
Contact <sip:gw+asterisk@52.67.78.120:5090;transport=udp;gw=asterisk>
Exten FreeSWITCH
To sip:FreeSWITCH@192.241.203.21:5090
Proxy sip:192.241.203.21:5090
Context public
Expires 3600
Freq 3600
Ping 0
PingFreq 0
PingTime 0.00
PingState 0/0/0
State NOREG
Status UP
Uptime 290s
CallsIN 0
CallsOUT 2
FailedCallsIN 0
FailedCallsOUT 0
sofia status profile external
=================================================================================================
Name external
Domain Name N/A
Auto-NAT false
DBName sofia_reg_external
Pres Hosts
Dialplan XML
Context public
Challenge Realm auto_to
RTP-IP 172.31.3.166
Ext-RTP-IP 52.67.78.120
SIP-IP 172.31.3.166
Ext-SIP-IP 52.67.78.120
URL sip:mod_sofia@52.67.78.120:5090
BIND-URL sip:mod_sofia@52.67.78.120:5090;maddr=172.31.3.166;transport=udp,tcp
HOLD-MUSIC local_stream://moh
OUTBOUND-PROXY N/A
CODECS IN OPUS,G722,PCMU,PCMA,VP8
CODECS OUT OPUS,G722,PCMU,PCMA,VP8
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 true
CALLS-IN 0
FAILED-CALLS-IN 0
CALLS-OUT 2
FAILED-CALLS-OUT 1
REGISTRATIONS 0
Mod_spandsp
>module_exists mod_spandsp
true
>show applications
spandsp_detect_tdd,Detect TDD data,,mod_spandsp
spandsp_inject_tdd,Send TDD data,,mod_spandsp
spandsp_send_tdd,Send TDD data,,mod_spandsp
spandsp_start_dtmf,Detect dtmf,,mod_spandsp
spandsp_start_fax_detect,start fax detect,<app>[ <arg>][ <timeout>][ <tone_type>],mod_spandsp
spandsp_start_tone_detect,Start background tone detection with cadence,<name>,mod_spandsp
spandsp_stop_detect_tdd,stop sending tdd,,mod_spandsp
spandsp_stop_dtmf,stop inband dtmf,,mod_spandsp
spandsp_stop_fax_detect,stop fax detect,,mod_spandsp
spandsp_stop_inject_tdd,stop sending tdd,,mod_spandsp
spandsp_stop_tone_detect,Stop background tone detection with cadence,,mod_spandsp
Thanks