Problem is solved after
externip=<external NAT IP> addition on aserisk.
31.10.2013 0:02, "Гришин В.С." пишет:
Call in direction from IP Phone to
PBX works fine.
30.10.2013 22:57, "Гришин В.С." пишет:
the correct scheme is PBX---E1--Cisco
2811-----VoIP SIP-----FreeSWITCH-----VoIP
SIP----NAT----asterisk(1.8.20)---IP
Phone
Asterisk IP is 162.48.13.26
(External IP NAT
)
30.10.2013 22:36, "Гришин В.С." пишет:
the scheme is
PBX---E1--Cisco
2811-----VoIP SIP-----FreeSWITCH-----VoIP
SIP---asterisk(1.8.20)---IP
Phone
FreeSWITCH IP is
191.21.19.12
Asterisk IP is 162.48.13.26
Call comes to FreeSWITCH from a Cisco 2811.
SIP session between freeswitch
and asterisk(1.8.20)
|Time | 191.21.19.12
|
| | | 162.48.13.26
|
|21.048754| INVITE SDP (g711A CN) |SIP
From: "91295" <sip:9195@191.21.19.12
To:<sip:9647@162.48.13.26
| |(5065) ------------------> (5060) |
|21.048758| INVITE SDP (g711A CN) |SIP
From: "91295" <sip:9195@191.21.19.12
To:<sip:9647@162.48.13.26
| |(5065) ------------------> (5060) |
|21.065794| 100 Trying| |SIP
Status
| |(5065) <------------------ (5060) |
|21.122406| 180 Ringing |SIP
Status
| |(5065) <------------------ (5060) |
|21.143580| RTP (g711A) |RTP
Num packets:50 Duration:0.922s SSRC:0xB5320A6
| |(21866) <------------------ (12772) |
|22.282684| 180 Ringing |SIP
Status
| |(5065) <------------------ (5060) |
|22.296762| 181 Call is being forwarded |SIP
Status
| |(5065) <------------------ (5060) |
|27.558268| 180 Ringing |SIP
Status
| |(5065) <------------------ (5060) |
|35.039883| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|35.307222| RTP (g711A) |RTP
Num packets:1490 Duration:29.773s SSRC:0x530CAB37
| |(21866) <------------------ (12772) |
|35.476614| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|35.532995| RTP (g711A) |RTP
Num packets:2966 Duration:29.680s SSRC:0x308FC816
| |(21866) ------------------> (12772) |
|36.436846| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|38.282686| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|42.025399| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|45.637280| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|49.343115| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|53.119940| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|56.907800| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|60.791453| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|64.606730| 200 OK SDP (g711A) |SIP
Status
| |(5065) <------------------ (5060) |
|65.219392| BYE | |SIP
Request
| |(5065) <------------------ (5060) |
|65.234354| 200 OK | |SIP
Status
| |(5065) ------------------> (5060) |
|65.234359| 200 OK | |SIP
Status
| |(5065) ------------------> (5060) |
It
is visible that there is no package with ACK
confirmation from a FreeSWITCH
to Asterisk. Asterisk sends several
times SIP OK on a freeswitch,
and without having waited confirmations (ACK)
after 23 seconds sends SIP BY on a freeswitch
and on IP Phone. After
that call stops. Before disconnection subscribers hear each
other of 29 (+-) seconds
In tcpdump
between Cisco and FreeSWITCH
it is visible that the SIP ACK packet
from Cisco arrives on a freeswitch.
In what
place the SIP ACK in a FreeSWITCH disappeared?
What to me to analyse in SIP session?
Maybe someone solved a similar problem?
please,
Help!
Vladislav Grishin
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Vladislav Grishin
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Vladislav Grishin
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com
Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org