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

[Daniel Neubert]

I think solution to this issue might be  to configure

qualify=yes

in sip.conf in Asterisk. I'll monitor this solution for a few days but 
it looks good for now.

I'll try to dismiss the ping feature but I like to have it's events via 
ESL interface...

Best regards / Mit freundlichen Grüßen,
Daniel

On 28.07.2010 21:17, Anthony Minessale wrote:
> 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 <mailto: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
>>>     <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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>
> -- 
> Anthony Minessale II
>
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