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

[Daniel Neubert]

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 
>> <mailto: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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