[Freeswitch-users] Call problems

[Jonas Gauffin]

I've added some checks to prevent the problem, but I wont submit a
patch since I really doesnt know whats expected to be in
tech_pvt->invite_contact.

But after doing a small fix, I got everything working.
And transfer is working fine now. You can close the issue in jira.

Please take a look at SFSIP-28, it's a showstopper for me. I cannot
answer any external calls, they are hungup when I do.

On 10/26/07, Jonas Gauffin <jonas.gauffin at gmail.com> wrote:
> 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
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > 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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