[Freeswitch-users] FreeSWITCH re-INVITE breaks CDRs???

[Joel Serrano]

Never mind, I found out how to enable the sip-traces on the logfile.

As soon as it happens again I will open a JIRA.

Thanks.

On Thu, Sep 22, 2016 at 9:55 AM, Joel Serrano <joel at gogii.net> wrote:

> HI Ken,
>
> 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.
>
> How can I enable siptrace in the logfiles?
>
> I will open a JIRA as soon as I find a case that happened after enabling
> DEBUG logs and siptrace.
>
> Thanks!
> Joel.
>
> On Wed, Sep 21, 2016 at 7:47 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> And if you think this is a bug open anjora and attach a full unedited
>> debug log with sip tracing and the xml cdr.
>>
>> Sent from my iPhone
>>
>> > On Sep 21, 2016, at 8:31 PM, Joel Serrano <joel at gogii.net> wrote:
>> >
>> > Hi,
>> >
>> > As I understand, freeswitch can handle re-invites. Ok till here, but,
>> is there any scenario where freeswitch will send a re-invite proactively ?
>> >
>> > I'm facing the following problem:
>> >
>> > Inbound call, from a random person to one of our numbers.
>> > Call from: +1706288XXXX (random caller)
>> > Call to: +1407602XXXX (our number)
>> >
>> >
>> > This is the diagram:
>> >
>> > <Screen Shot 2016-09-21 at 18.15.16.png>
>> >
>> >
>> > 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.
>> >
>> > 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.
>> >
>> > I use a script to pull full XML cdrs from our database... The CDR for
>> this call has the following:
>> >
>> > $ ./get_cdr_by_uuid.py '1367e1db-2df4-4c97-8b23-304f85493315' | grep
>> -i -e sip_from_user -e sip_to_user
>> >     <sip_from_user_stripped>17062886161</sip_from_user_stripped>
>> >     <sip_from_user>%2B14076027616</sip_from_user>
>> >     <sip_to_user>%2B17062886161</sip_to_user>
>> > $
>> >
>> > 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?
>> >
>> >
>> > I don't know if what can be happening.
>> >
>> > This is happening on FS v1.6.10.
>> >
>> > I have servers on older versions, I'm trying to reproduce to see if it
>> is related to that specific version of FS.
>> >
>> > I'm also trying to get debug logs, which I currently don't have
>> (because they are disabled in prod).
>> >
>> >
>> > Any initial ideas?
>> >
>> > 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.
>> >
>> >
>> > Thanks,
>> > Joel.
>> >
>> >
>> >
>> >
>> >
>> > ____________________________________________________________
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>>
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