[Freeswitch-users] Call problems

[Jonas Gauffin]

A NULL pointer is returned from "d_contact =
sofia_glue_get_url_from_contact(tech_pvt->invite_contact, 1);" in
sofia_glue.c, line 589

That results in that switch_stristr on the line below tries to work
with a NULL pointer and crashes.

tech_pvt->invite_contact contains "sip:mod_sofia at 192.168.1.102:5060" here.

(I love Visual Studio remote debugging ;) )

On 10/25/07, Anthony Minessale <anthmct at yahoo.com> wrote:
>
> please retest with latest trunk as a few issues were fixed with transfers
> today.
>
>
> Be aware that the transfers work best with bridged calls.
> It sounds like that is what you are doing so it should be ok but there are a
> few things to consider when calling an extension that is just "playback" for
> example.
>
> Say you call an extension that has playback foo.wav:
>
> Technically the file is playing on the inbound leg and this is considered a
> 1 legged call.
>
> so if you blind transfer this extension there is nothing to transfer cos
> there is only 1 leg so in this case we will not allow it.
>
> if you attended transfer this extension, we will transfer the new channel
> you called to the same extension as the one you were on and hangup the
> original so the playback will start over for the called party.
>
>
> if you want to call and ivr or playfile extension and then transfer it you
> have to make a hairpin call back into the system.
>
> so instead of making 1000 lead to playback foo.wav
> make it lead to sofia/default/playback-foo at same.box
>
> then you have 2 legs and it's possible to transfer the playback.
>
>
>
>
>
>
>
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
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>
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> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> ----- Original Message ----
> From: Jonas Gauffin <jonas.gauffin at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Thursday, October 25, 2007 6:55:45 AM
> Subject: Re: [Freeswitch-users] Call problems
>
>  Ok. I've created a jira entry.
>
> On 10/25/07, Michael Jerris <mike at jerris.com> wrote:
> > We would need a full trace from freeswitch including the TPORT_LOG to
> > determine that.
> >
> >
> > On 10/25/07, Jonas Gauffin < jonas.gauffin at gmail.com> wrote:
> > >
> > > Hello
> > >
> > > I have three phones, all registered with FreeSwitch.
> > >
> > > I did the following:
> > > a) Called from phone A to phone B
> > > b) B picked up
> > > c) A put B on hold
> > > c) A called phone C
> > > d) C picked up
> > > e) Pressed transfer on A's phone
> > >
> > > Both B and C got disconnected instead of being connected to each other.
> > >
> > > I did another variant:
> > > A calls B, B calls C, B presses transfer.
> > > A things that he is still connected, B and C gets disconnected.
> > >
> > > Are these problems in my phones, or are they FS problems?
> > >
> > > //Jonas
> > >
> > > _______________________________________________
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