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. Mike On 10/26/07, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
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@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@192.168.1.102:5060" here.
(I love Visual Studio remote debugging ;) )
On 10/25/07, Anthony Minessale <anthmct@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@same.box
then you have 2 legs and it's possible to transfer the playback.
Anthony Minessale II
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----- Original Message ---- From: Jonas Gauffin <jonas.gauffin@gmail.com> To: freeswitch-users@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@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@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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