[Freeswitch-users] ACK disappeared from SIP session, Help

["Гришин В.С."]

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 at 191.21.19.12 To:<sip:9647 at 162.48.13.26
>> |         |(5065)   ------------------>  (5060)   |
>> |21.048758|         INVITE SDP (g711A CN)         |SIP From: "91295" 
>> <sip:9195 at 191.21.19.12 To:<sip:9647 at 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 at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131031/0666425a/attachment-0001.html