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I'll keep that in mind. If more information is needed to get into
the bottom of this, I will happily oblige. Thanks for helping.<br>
<br>
On 04/06/2011 03:09 AM, Michael Collins wrote:
<blockquote
cite="mid:BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com"
type="cite">I'll have to defer to those more experienced than I in
such matters. However, I can offer two tips:
<div><br>
</div>
<div>#1 - turn off the crazy sofia debugging - it's just noise.
All you need to do to enable SIP trace is "sofia global siptrace
on"</div>
<div>#2 - when you pastebin the console output use the FreeSWITCH
log syntax highlighting - it makes it *much* easier to see
what's going on.</div>
<div><br>
</div>
<div>-MC<br>
<br>
<div class="gmail_quote">On Mon, Apr 4, 2011 at 10:51 PM, Joegen
E. Baclor <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:joegen@opensipstack.org">joegen@opensipstack.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000000"> Hi Michael,<br>
<br>
I have pasted both working and none working logs on
pastebin.<br>
<br>
FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)<br>
working: <a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/16008"
target="_blank">http://pastebin.freeswitch.org/16008</a><br>
not working: <a moz-do-not-send="true"
href="http://pastebin.freeswitch.org/16009"
target="_blank">http://pastebin.freeswitch.org/16009</a><br>
<br>
The call flow for the working call is<br>
UA1 -> (FSBridgeDialPlan) -> (SIP-Loopback) ->
(FSIVRApp)<br>
FSIVRApp knows the uuid of the bridge call. Pressing # on
the IVR results to a uuid_deflect on the bridged channel.
This works and call successfully transfers to the new
destination.<br>
<br>
The call flow for the none working call is<br>
<br>
1. UA1 -> UA2 is in conversation<br>
2. UA1 puts UA2 on hold<br>
<br>
-- start of FS interaction here --<br>
<br>
3. UA1 -> (FSBridgeDialPlan) -> (SIP-Loopback)
-> (FSIVRApp) (on line 2)<br>
4. UA1 sends REFER (replacing its call with UA2) to
FSBridgeDialPlan.<br>
5. Flow is now UA2 -> ([REPLACED]FSBridgeDialPlan)
-> (SIP-Loopback) -> (FSIVRApp) <br>
6. UA2 presses #.<br>
7. IVRApp performs uuid_deflect on FSBridgeDialPlan.<br>
8. FSBridgeDialPlan drops call (no REFER is done)<br>
<br>
Thanks for your help.<br>
<font color="#888888"> <br>
Joegen</font>
<div>
<div class="h5"><br>
<br>
On 04/05/2011 12:35 PM, Michael Collins wrote:
<blockquote type="cite">What do you see on the console
when you try this? A console debug log with siptrace
would go a long way toward figuring out what is
happening.
<div><br>
</div>
<div>-MC<br>
<br>
<div class="gmail_quote">On Mon, Apr 4, 2011 at
9:27 PM, Joegen E. Baclor <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:joegen@opensipstack.org"
target="_blank">joegen@opensipstack.org</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:
0pt 0pt 0pt 0.8ex; border-left: 1px solid
rgb(204, 204, 204); padding-left: 1ex;">Hi
List,<br>
<br>
I have a scenario where a bridged call has
been replaced due to a<br>
consultative transfer. This works pretty well
and audio is<br>
bidirectional. I have the original uuid of
the call in a var<br>
somewhere. The trouble begins when I
uuid_deflect the bridged call once<br>
again to attempt another transfer. Sofia
disconnects the channel. I am<br>
using the original uuid of the call (uuid
prior to replaces). Is this<br>
the right way of doing it?<br>
<br>
Joegen<br>
<br>
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</blockquote>
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