[Freeswitch-users] Keep alive for SIP trunk between Asterisk and Freeswitch

[Anthony Minessale]

you could just omit the ping param if asterisk can't operate properly with
that feature.


On Wed, Jul 28, 2010 at 8:52 AM, Daniel Neubert <daniel.neubert at solomo.de>wrote:

>  Now I have a trace from Freeswitch log:
>
> 2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down!
> 2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A
> [CS_NEW]
> 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 ()
> Running State Change CS_DESTROY
> 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A)
> State DESTROY
> 2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY
> 2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440 (N/A)
> State DESTROY going to sleep
> 2010-07-28 15:49:26.640905 [ERR] switch_ivr_originate.c:2623 Cannot create
> outgoing channel of type [sofia] cause: [NETWORK_OUT_OF_ORDER]
> 2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431 Originate
> Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER]
> 2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed.
> Cause: NETWORK_OUT_OF_ORDER
> 2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309
> (sofia/internal/test01 at voip-test) Callstate Change EARLY -> HANGUP
> 2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
> sofia/internal/test01 at voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER]
>
> Directly after that I called in from PSTN -> SS7 -> Asterisk so the gateway
> came up again and outbound call was possible.
>
>  Best regards / Mit freundlichen Grüßen,
> Daniel
>
> On 28.07.2010 11:33, Daniel Neubert wrote:
>
> The call fails because the desired gateway is down.
>
> Logs are not available at the moment and issue cannot be reproduced on
> demand. I'll take logs as soon as this occurs again.
>
> Best regards / Mit freundlichen Grüßen,
> Daniel
>
>
> On 28.07.2010 10:08, Steven Ayre wrote:
>
> Where & why does the call fail?
>
> Do you have any log file output?
>
> -Steve
>
>
>
>
> On 28 July 2010 08:25, Daniel Neubert <daniel.neubert at solomo.de> wrote:
>
>> Hi,
>>
>> we've set  up a SIP trunk between Asterisk (used as MediaGateway to
>> SS7-Network for PSTN access) and Freeswitch.
>>
>> Everything works fine except one "little" issue: If there have been no
>> calls using the SIP trunk it becomes unuseable from Freeswitch side.
>>
>> PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch <- SIP/RTP ->
>> VoIP Clients
>>
>> If a user tries to originate the call that is routed to one of our
>> MediaGateways while SIP trunk is "stale", the call will fail. The trunk
>> can be made available again by calling in via PSTN -> Asterisk ->
>> SIP-Trunk
>>
>> This is our gateway configuration (tried using low values for
>> expire-seconds, ping and retry-seconds to keep the connection up:
>>
>> <gateway name="voip-int-test">
>> <param name="username" value="voip-ext-test"/>
>> <param name="password" value="freeswitch"/>
>> <param name="proxy" value="172.31.45.43"/>
>> <param name="register" value="false"/>
>> <param name="expire-seconds" value="15"/>
>> <param name="ping" value="5"/>
>> <param name="retry-seconds" value="5"/>
>> <param name="context" value="default"/>
>> <param name="apply-inbound-acl" value="voip-int-test"/>
>> <param name="caller-id-in-from" value="true"/>
>> </gateway>
>>
>>
>>
>> --
>>
>> Best regards / Mit freundlichen Grüßen,
>> Daniel
>>
>>
>>
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>>
>
>
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>


-- 
Anthony Minessale II

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