If you have full control and access to both the FS and Asterisk, you could "not register" either of them to the other at all. You could just add FS IP as a "peer" in Asterisk. And, in FS, you could just allow calls from the Asterisk based on IP.<div>
<br></div><div>Assuming, of course, no NAT is involved between FS and Asterisk.</div><div><br></div><div>Regards</div><div>HASSAN</div><div><br></div><div><br><br><div class="gmail_quote">On Thu, Jul 29, 2010 at 01:17, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">you could just omit the ping param if asterisk can't operate properly with that feature.<div><div><div></div><div class="h5">
<br><br><div class="gmail_quote">On Wed, Jul 28, 2010 at 8:52 AM, Daniel Neubert <span dir="ltr"><<a href="mailto:daniel.neubert@solomo.de" target="_blank">daniel.neubert@solomo.de</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Verdana">Now I have a trace from Freeswitch
log:<br>
<br>
2010-07-28 15:49:26.630733 [ERR] mod_sofia.c:3674 Gateway is down!<br>
2010-07-28 15:49:26.630733 [NOTICE] mod_sofia.c:3984 Close Channel N/A
[CS_NEW]<br>
2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:430 ()
Running State Change CS_DESTROY<br>
2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440
(N/A) State DESTROY<br>
2010-07-28 15:49:26.640905 [DEBUG] mod_sofia.c:358 N/A SOFIA DESTROY<br>
2010-07-28 15:49:26.640905 [DEBUG] switch_core_state_machine.c:440
(N/A) State DESTROY going to sleep<br>
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]<br>
2010-07-28 15:49:26.642906 [DEBUG] switch_ivr_originate.c:3431
Originate Resulted in Error Cause: 38 [NETWORK_OUT_OF_ORDER]<br>
2010-07-28 15:49:26.642906 [INFO] mod_dptools.c:2393 Originate Failed.
Cause: NETWORK_OUT_OF_ORDER<br>
2010-07-28 15:49:26.644905 [DEBUG] switch_channel.c:2309
(sofia/internal/test01@voip-test) Callstate Change EARLY -> HANGUP<br>
2010-07-28 15:49:26.644905 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/test01@voip-test [CS_EXECUTE] [NETWORK_OUT_OF_ORDER]<br>
<br>
Directly after that I called in from PSTN -> SS7 -> Asterisk so
the gateway came up again and outbound call was possible.<br>
<br>
</font></font><div>
<pre cols="72">Best regards / Mit freundlichen Grüßen,
Daniel
</pre></div><div><div></div><div>
On 28.07.2010 11:33, Daniel Neubert wrote:
<blockquote type="cite">
The call fails because the desired gateway is down. <br>
<br>
Logs are not available at the moment and issue cannot be reproduced on
demand. I'll take logs as soon as this occurs again.<br>
<pre cols="72">Best regards / Mit freundlichen Grüßen,
Daniel
</pre>
On 28.07.2010 10:08, Steven Ayre wrote:
<blockquote type="cite">Where & why does the call fail?<br>
<br>
Do you have any log file output?<br>
<br>
-Steve<br>
<br>
<br>
<br>
<br>
<div class="gmail_quote">On 28 July 2010 08:25, Daniel Neubert <span dir="ltr"><<a href="mailto:daniel.neubert@solomo.de" target="_blank">daniel.neubert@solomo.de</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="border-left:1px solid rgb(204, 204, 204);margin:0pt 0pt 0pt 0.8ex;padding-left:1ex">Hi,<br>
<br>
we've set up a SIP trunk between Asterisk (used as MediaGateway to<br>
SS7-Network for PSTN access) and Freeswitch.<br>
<br>
Everything works fine except one "little" issue: If there have been no<br>
calls using the SIP trunk it becomes unuseable from Freeswitch side.<br>
<br>
PSTN <- SS7/ISUP -> Asterisk <- SIP Trunk -> Freeswitch
<- SIP/RTP -><br>
VoIP Clients<br>
<br>
If a user tries to originate the call that is routed to one of our<br>
MediaGateways while SIP trunk is "stale", the call will fail. The trunk<br>
can be made available again by calling in via PSTN -> Asterisk ->
SIP-Trunk<br>
<br>
This is our gateway configuration (tried using low values for<br>
expire-seconds, ping and retry-seconds to keep the connection up:<br>
<br>
<gateway name="voip-int-test"><br>
<param name="username" value="voip-ext-test"/><br>
<param name="password" value="freeswitch"/><br>
<param name="proxy" value="172.31.45.43"/><br>
<param name="register" value="false"/><br>
<param name="expire-seconds" value="15"/><br>
<param name="ping" value="5"/><br>
<param name="retry-seconds" value="5"/><br>
<param name="context" value="default"/><br>
<param name="apply-inbound-acl" value="voip-int-test"/><br>
<param name="caller-id-in-from" value="true"/><br>
</gateway><br>
<br>
<br>
<br>
--<br>
<br>
Best regards / Mit freundlichen Grüßen,<br>
Daniel<br>
<br>
<br>
<br>
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