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@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. 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@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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