[Freeswitch-users] Transfer attempt for a previously a replaced call fails
Joegen E. Baclor
joegen at opensipstack.org
Wed Apr 6 04:02:10 MSD 2011
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.
On 04/06/2011 03:09 AM, Michael Collins wrote:
> I'll have to defer to those more experienced than I in such matters.
> However, I can offer two tips:
>
> #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"
> #2 - when you pastebin the console output use the FreeSWITCH log
> syntax highlighting - it makes it *much* easier to see what's going on.
>
> -MC
>
> On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor
> <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>
> Hi Michael,
>
> I have pasted both working and none working logs on pastebin.
>
> FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
> working: http://pastebin.freeswitch.org/16008
> not working: http://pastebin.freeswitch.org/16009
>
> The call flow for the working call is
> UA1 -> (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
> 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.
>
> The call flow for the none working call is
>
> 1. UA1 -> UA2 is in conversation
> 2. UA1 puts UA2 on hold
>
> -- start of FS interaction here --
>
> 3. UA1 -> (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
> (on line 2)
> 4. UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
> 5. Flow is now UA2 -> ([REPLACED]FSBridgeDialPlan) ->
> (SIP-Loopback) -> (FSIVRApp)
> 6. UA2 presses #.
> 7. IVRApp performs uuid_deflect on FSBridgeDialPlan.
> 8. FSBridgeDialPlan drops call (no REFER is done)
>
> Thanks for your help.
>
> Joegen
>
>
> On 04/05/2011 12:35 PM, Michael Collins wrote:
>> 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.
>>
>> -MC
>>
>> On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor
>> <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>>
>> Hi List,
>>
>> I have a scenario where a bridged call has been replaced due to a
>> consultative transfer. This works pretty well and audio is
>> bidirectional. I have the original uuid of the call in a var
>> somewhere. The trouble begins when I uuid_deflect the
>> bridged call once
>> again to attempt another transfer. Sofia disconnects the
>> channel. I am
>> using the original uuid of the call (uuid prior to replaces).
>> Is this
>> the right way of doing it?
>>
>> Joegen
>>
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