Jonas, can you catch me online on irc to discuss this? If not, can we arrange a phone call or IM conversation to troubleshoot this and get the right patch in tree.<br><br>Mike<br><br><div><span class="gmail_quote">On 10/26/07,
<b class="gmail_sendername">Jonas Gauffin</b> <<a href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I've added some checks to prevent the problem, but I wont submit a<br>patch since I really doesnt know whats expected to be in<br>tech_pvt->invite_contact.<br><br>But after doing a small fix, I got everything working.
<br>And transfer is working fine now. You can close the issue in jira.<br><br>Please take a look at SFSIP-28, it's a showstopper for me. I cannot<br>answer any external calls, they are hungup when I do.<br><br>On 10/26/07, Jonas Gauffin <
<a href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>> wrote:<br>> A NULL pointer is returned from "d_contact =<br>> sofia_glue_get_url_from_contact(tech_pvt->invite_contact, 1);" in<br>
> sofia_glue.c, line 589<br>><br>> That results in that switch_stristr on the line below tries to work<br>> with a NULL pointer and crashes.<br>><br>> tech_pvt->invite_contact contains "sip:mod_sofia@192.168.1.102
:5060" here.<br>><br>> (I love Visual Studio remote debugging ;) )<br>><br>> On 10/25/07, Anthony Minessale <<a href="mailto:anthmct@yahoo.com">anthmct@yahoo.com</a>> wrote:<br>> ><br>> > please retest with latest trunk as a few issues were fixed with transfers
<br>> > today.<br>> ><br>> ><br>> > Be aware that the transfers work best with bridged calls.<br>> > It sounds like that is what you are doing so it should be ok but there are a<br>> > few things to consider when calling an extension that is just "playback" for
<br>> > example.<br>> ><br>> > Say you call an extension that has playback foo.wav:<br>> ><br>> > Technically the file is playing on the inbound leg and this is considered a<br>> > 1 legged call.
<br>> ><br>> > so if you blind transfer this extension there is nothing to transfer cos<br>> > there is only 1 leg so in this case we will not allow it.<br>> ><br>> > if you attended transfer this extension, we will transfer the new channel
<br>> > you called to the same extension as the one you were on and hangup the<br>> > original so the playback will start over for the called party.<br>> ><br>> ><br>> > if you want to call and ivr or playfile extension and then transfer it you
<br>> > have to make a hairpin call back into the system.<br>> ><br>> > so instead of making 1000 lead to playback foo.wav<br>> > make it lead to <a href="mailto:sofia/default/playback-foo@same.box">
sofia/default/playback-foo@same.box</a><br>> ><br>> > then you have 2 legs and it's possible to transfer the playback.<br>> ><br>> ><br>> ><br>> ><br>> ><br>> ><br>> >
<br>> > Anthony Minessale II<br>> ><br>> > FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>> > ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a>
<br>> ><br>> > AIM: anthm<br>> > <a href="mailto:MSN:anthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>> > GTALK/JABBER/PAYPAL:<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com
</a><br>> > IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br>> ><br>> > FreeSWITCH Developer Conference<br>> > <a href="mailto:sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org
</a><br>> > <a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>> > <a href="mailto:googletalk:conf+888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org
</a><br>> > pstn:213-799-1400<br>> ><br>> ><br>> > ----- Original Message ----<br>> > From: Jonas Gauffin <<a href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>><br>> > To:
<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>> > Sent: Thursday, October 25, 2007 6:55:45 AM<br>> > Subject: Re: [Freeswitch-users] Call problems<br>> >
<br>> > Ok. I've created a jira entry.<br>> ><br>> > On 10/25/07, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> wrote:<br>> > > We would need a full trace from freeswitch including the TPORT_LOG to
<br>> > > determine that.<br>> > ><br>> > ><br>> > > On 10/25/07, Jonas Gauffin < <a href="mailto:jonas.gauffin@gmail.com">jonas.gauffin@gmail.com</a>> wrote:<br>> > > >
<br>> > > > Hello<br>> > > ><br>> > > > I have three phones, all registered with FreeSwitch.<br>> > > ><br>> > > > I did the following:<br>> > > > a) Called from phone A to phone B
<br>> > > > b) B picked up<br>> > > > c) A put B on hold<br>> > > > c) A called phone C<br>> > > > d) C picked up<br>> > > > e) Pressed transfer on A's phone
<br>> > > ><br>> > > > Both B and C got disconnected instead of being connected to each other.<br>> > > ><br>> > > > I did another variant:<br>> > > > A calls B, B calls C, B presses transfer.
<br>> > > > A things that he is still connected, B and C gets disconnected.<br>> > > ><br>> > > > Are these problems in my phones, or are they FS problems?<br>> > > ><br>> > > > //Jonas
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