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

[David Ponzone]

Do you see the SIP OPTIONS sent by FS to Asterisk every 5 seconds ?
Do you see Asterisk's replies ?
What is it ? (depending on what Asterisk replies as SIP Error code, FS  
could decide to down the gateway)
Normally, Asterisk should reply 404 or 200.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
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Service Client IPeva
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Le 28/07/2010 à 11:33, Daniel Neubert a écrit :

> 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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