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 href="mailto:joegen@opensipstack.org">joegen@opensipstack.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;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 href="http://pastebin.freeswitch.org/16008" target="_blank">http://pastebin.freeswitch.org/16008</a><br>
not working: <a 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></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 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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