[Freeswitch-users] Freeswitch don't send keepalive
Tanguy
phenix at vfemail.net
Mon Jan 26 00:43:14 MSK 2015
Hello
I'm a asterisk user, and i would like to try freeswitch for having a
real multi tenant system. I configured FS with fusionpbx. My FS is
installed on a VPS ( has a public IP ), my extensions are behind a nat
router.
After few minutes my phone became unreachable for inbound call. After a
outbound call, the phone is again reachable for few minutes.
I captured the traffic on server side with tcpdump, unlike my asterisk
server, freeswitch never send to the phone a keepalive packet ( SIP
OPTIONS )
I added theses options in sip_profiles/internal.xml
<param name="aggressive-nat-detection" value="true"/>
<param name="enable-timer" value="false"/>
<param name="NDLB-received-in-nat-reg-contact" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="NDLB-broken-auth-hash" value="true"/>
<param name="sip-force-expires" value="240"/>
<param name="all-reg-options-ping" value="true"/>
<param name="registration-thread-frequency" value="30"/>
<param name="unregister-on-options-fail" value="true"/>
<param name="nat-options-ping" value="true"/>
Please note the strange expiry delay. After being powered off, the phone
stays registered.
freeswitch at internal> sofia status profile internal reg
Registrations:
=================================================================================================
Call-ID: 24ef-c0a80101-5-1 at 192.168.0.10
User: 7082 at mydomain
Contact: ""
<sip:7082 at 192.168.0.10:5090;transport=udp;user=phone;fs_nat=yes;fs_path=sip%3A7082%4086.201.x.x%3A5090%3Btransport%3Dudp%3Buser%3Dphone>
Agent: THOMSON ST2030 hw5 fw2.76 00-14-7F-00-4C-1A
Status: Registered(UDP-NAT)(unknown) EXP(1907-10-11 18:28:34)
EXPSECS(909113102)
Ping-Status: Reachable
Host: vpsxxxxxx.ovh.net
IP: 86.201.x.x
Port: 5090
Auth-User: 7082
Auth-Realm: mydomain
MWI-Account: 7082 at mydomain
Any idea ?
Best regards
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