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