[Freeswitch-users] Call dropping after 32 seconds

Tue Feb 23 10:37:39 MSK 2016

Hi,

1) You have very complex set-up and I doubt that you need it.

2) As far as you have user with ip x.x.x.174 and opensips server with same
ip x.x.x.174 it very hard to debug. So I propose you to send new log where
will be difference between user ip and opensips IP.

3) If you have possibility, try to register directly with a user to x.x.x.3
gateway and check if same issue still exists, if there is no such issue
anymore, then thee is definitely issue in your opensips x.x.x.174 and
freeswitch x.x.x.166. My point here is that you need to isolate issue and
to understand what part of your set-up works as expected and what is faulty.

With kind regards,

Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:

> Thanks for the reply.
>
> Got your point about NATing issue and no response of 200 OK and as a
> resoult ACK Timeout.
> So, now to resolve the issue, if you can assist, what could be the
> possible fixies?
> From where can i start? where to look?
>
> Thanks.
>
> --
> Thanks,
> Rutu Patel
>
> <http://www.inextrix.com>
>
> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>
>> You have lots of 200s to the user since it's waiting for an ACK and keeps
>> retrying, but for whatever network reason (router... sip alg?), it isn't
>> getting one, so it triggers a timer to stop the call.
>>
>>
>> -Avi Marcus
>> BestFone
>>
>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Having issue of call dropping after 32 seconds, here are the details-
>>>
>>> x.x.x.174: opensips server
>>> x.x.x.166: freeswitch server
>>> x.x.x.3:     another opensips server which is registered as gateway on
>>> above freeswitch server
>>> x.x.x.6: freeswitch server
>>> x.x.x.47:  server through which the user is registered
>>> I am trying to call from xxxx9 to xxxxxxx29858
>>> xxxxxxx00181 is caller-id name and caller-id number
>>>
>>> Call flow is like this:
>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>
>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>
>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>> 2) Call hit the freeswitch server x.x.x.166
>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>> gateway x.x.x.3
>>> But after 32 seconds call is dropped,
>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>> issue.
>>> Here I have attached the file with sip logs, you can observer from the
>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>
>>> What is wrong here? Any help would be appreciated here.
>>>
>>> Here is the file with sip logs
>>> --
>>> Thanks,
>>> Rutu Patel
>>> <http://www.inextrix.com>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>

Jurijs

2016-02-23 7:51 GMT+02:00 Rutu Patel <rutu.patel at inextrix.com>:

> Thanks for the reply.
>
> Got your point about NATing issue and no response of 200 OK and as a
> resoult ACK Timeout.
> So, now to resolve the issue, if you can assist, what could be the
> possible fixies?
> From where can i start? where to look?
>
> Thanks.
>
> --
> Thanks,
> Rutu Patel
>
> <http://www.inextrix.com>
>
> On Mon, Feb 22, 2016 at 12:06 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
>> 5 second response: 32 seconds is a timer/[network/NAT] issue.
>>
>> You have lots of 200s to the user since it's waiting for an ACK and keeps
>> retrying, but for whatever network reason (router... sip alg?), it isn't
>> getting one, so it triggers a timer to stop the call.
>>
>>
>> -Avi Marcus
>> BestFone
>>
>> On Mon, Feb 22, 2016 at 8:19 AM, Rutu Patel <rutu.patel at inextrix.com>
>> wrote:
>>
>>> Hello,
>>>
>>> Having issue of call dropping after 32 seconds, here are the details-
>>>
>>> x.x.x.174: opensips server
>>> x.x.x.166: freeswitch server
>>> x.x.x.3:     another opensips server which is registered as gateway on
>>> above freeswitch server
>>> x.x.x.6: freeswitch server
>>> x.x.x.47:  server through which the user is registered
>>> I am trying to call from xxxx9 to xxxxxxx29858
>>> xxxxxxx00181 is caller-id name and caller-id number
>>>
>>> Call flow is like this:
>>> registered user -> x.x.x.166 (freeswitch server) -> x.x.x.174 (opensips
>>> server) -> x.x.x.3 (Gateway) -> x.x.x.6 (freeswitch server)
>>>
>>> Outbound-proxy is set to x.x.x.174 in Gateway configuration.
>>>
>>> 1) Call is initiated by the user(xxxx9) registered with host x.x.x.174
>>> 2) Call hit the freeswitch server x.x.x.166
>>> 3) After '180 Ringing' and '183 Session Progress' packet
>>> sending-receiving started between 'x.x.x.174' and 'x.x.x.166' through the
>>> gateway x.x.x.3
>>> But after 32 seconds call is dropped,
>>> Within 32 seconds audio is ok from both end so it should not be the RTP
>>> issue.
>>> Here I have attached the file with sip logs, you can observer from the
>>> file that, there are many '200 OK' from x.x.x.174 to x.x.x.166
>>> and at the end, x.x.x.174 sending 'ACK Timeout' to x.x.x.166 and then
>>> 'BYE' from x.x.x.166 to x.x.x.174 and call is dropped.
>>>
>>> What is wrong here? Any help would be appreciated here.
>>>
>>> Here is the file with sip logs
>>> --
>>> Thanks,
>>> Rutu Patel
>>> <http://www.inextrix.com>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://confluence.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://confluence.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.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
>
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