[Freeswitch-users] No tone detection in AWS

[Gonzalo Gasca Meza]

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

http://pastebin.com/bPx4xGCP

*FreeSWITCH (Version 1.6.9 -16-d574870 64bit)* is ready AWS FAILS

http://pastebin.com/ktnFYJ05


*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 at 192.241.203.21:5090>

Contact <sip:gw+asterisk at 52.67.78.120:5090;transport=udp;gw=asterisk>

Exten   FreeSWITCH

To      sip:FreeSWITCH at 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 at 52.67.78.120:5090

BIND-URL         sip:mod_sofia at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160809/283b67e9/attachment-0001.html