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<font size="-1"><font face="Verdana">I think solution to this issue
might be to configure<br>
<br>
qualify=yes<br>
<br>
in sip.conf in Asterisk. I'll monitor this solution for a few days but
it looks good for now. <br>
<br>
I'll try to dismiss the ping feature but I like to have it's events via
ESL interface... <br>
<br>
</font></font>
<pre class="moz-signature" cols="72">Best regards / Mit freundlichen Grüßen,
Daniel
</pre>
On 28.07.2010 21:17, Anthony Minessale wrote:
<blockquote
cite="mid:AANLkTikMo3yHL+JR8zB67TRSj8CVm-3nJ4QB3TR52voa@mail.gmail.com"
type="cite">you could just omit the ping param if asterisk can't
operate properly with that feature.
<div><br>
<br>
<div class="gmail_quote">On Wed, Jul 28, 2010 at 8:52 AM, Daniel
Neubert <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:daniel.neubert@solomo.de">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;">
<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 class="im">
<pre cols="72">Best regards / Mit freundlichen Grüßen,
Daniel
</pre>
</div>
<div>
<div class="h5">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 moz-do-not-send="true"
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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