<div dir="ltr">HI Ken, <div><br></div><div>I have tried to enable sip-trace (sofia global siptrace on) but although I see it in console, it doen't get added to logfile.</div><div><br></div><div>How can I enable siptrace in the logfiles?</div><div><br></div><div>I will open a JIRA as soon as I find a case that happened after enabling DEBUG logs and siptrace.</div><div><br></div><div>Thanks!</div><div>Joel.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Sep 21, 2016 at 7:47 PM, Ken Rice <span dir="ltr"><<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">And if you think this is a bug open anjora and attach a full unedited debug log with sip tracing and the xml cdr.<br>
<span class="im HOEnZb"><br>
Sent from my iPhone<br>
<br>
> On Sep 21, 2016, at 8:31 PM, Joel Serrano <<a href="mailto:joel@gogii.net">joel@gogii.net</a>> wrote:<br>
><br>
</span><span class="im HOEnZb">> Hi,<br>
><br>
> As I understand, freeswitch can handle re-invites. Ok till here, but, is there any scenario where freeswitch will send a re-invite proactively ?<br>
><br>
> I'm facing the following problem:<br>
><br>
> Inbound call, from a random person to one of our numbers.<br>
> Call from: +1706288XXXX (random caller)<br>
> Call to: +1407602XXXX (our number)<br>
><br>
><br>
> This is the diagram:<br>
><br>
</span><span class="im HOEnZb">> <Screen Shot 2016-09-21 at 18.15.16.png><br>
><br>
><br>
</span><span class="im HOEnZb">> For some reason, after X amount of time, FreeSWITCH sends out an INVITE. I don't understand the reason. I have checked and there are no session-timers in the initial request that could trigger it.<br>
><br>
> The problem is not the re-invite, the problem is that the CDR for this call is messed up, because the sip_from_user ends up being the callee, and the sip_to_user ends up being the caller.<br>
><br>
> I use a script to pull full XML cdrs from our database... The CDR for this call has the following:<br>
><br>
> $ ./get_cdr_by_uuid.py '1367e1db-2df4-4c97-8b23-<wbr>304f85493315' | grep -i -e sip_from_user -e sip_to_user<br>
> <sip_from_user_stripped><a href="tel:17062886161" value="+17062886161">17062886161</a></sip_<wbr>from_user_stripped><br>
> <sip_from_user>%2B14076027616<<wbr>/sip_from_user><br>
> <sip_to_user>%2B17062886161</<wbr>sip_to_user><br>
> $<br>
><br>
> As you can see, the sip_to_user, is the caller, (and this is probably because of the re-invite from FS to the client). Also, isn't it weird that the sip_from_user_stripped still has the original caller correctly, but the sip_from_user is updated with the original callee?<br>
><br>
><br>
> I don't know if what can be happening.<br>
><br>
> This is happening on FS v1.6.10.<br>
><br>
> I have servers on older versions, I'm trying to reproduce to see if it is related to that specific version of FS.<br>
><br>
> I'm also trying to get debug logs, which I currently don't have (because they are disabled in prod).<br>
><br>
><br>
> Any initial ideas?<br>
><br>
> I think this might be a config setting or something, if it seems like a bug or similar I'm happy to open a JIRA.<br>
><br>
><br>
> Thanks,<br>
> Joel.<br>
><br>
><br>
><br>
><br>
><br>
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</div></div></blockquote></div><br></div>