Michael Jerris napsal(a):
> This issue is now fixed in svn trunk:
>
> http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=8062
>
> fix hung channels when using respond app with 1xx or 2xx responses or on
> re-invite in proxy/bypass media with 1xx and 2xx responses
>
> Mike
>
Thank you ver
kokoska rokoska napsal(a):
kokoska rokoska napsal(a):
Anthony Minessale napsal(a):
That's all fine but the call leg does exist?
phoneA -- legA ---> FS <--- legB--- phoneB
if phoneB leaves and FS did not get a bye legA will never end so
naturally when you hangup
phoneA you will get a 20
Get on IRC. Is that possible?
/b
On Apr 8, 2008, at 2:04 PM, kokoska rokoska wrote:
>
> kokoska rokoska napsal(a):
>>
>>
>> Anthony Minessale napsal(a):
>>> That's all fine but the call leg does exist?
>>>
>>>
>>> phoneA -- legA ---> FS <--- legB--- phoneB
>>>
>>>
>>> if phoneB leaves and FS di
kokoska rokoska napsal(a):
>
>
> Anthony Minessale napsal(a):
>> That's all fine but the call leg does exist?
>>
>>
>> phoneA -- legA ---> FS <--- legB--- phoneB
>>
>>
>> if phoneB leaves and FS did not get a bye legA will never end so
>> naturally when you hangup
>> phoneA you will get a 200ok
Anthony Minessale napsal(a):
> That's all fine but the call leg does exist?
>
>
> phoneA -- legA ---> FS <--- legB--- phoneB
>
>
> if phoneB leaves and FS did not get a bye legA will never end so
> naturally when you hangup
> phoneA you will get a 200ok on it's leg.
>
> That is why I asked
That's all fine but the call leg does exist?
phoneA -- legA ---> FS <--- legB--- phoneB
if phoneB leaves and FS did not get a bye legA will never end so naturally
when you hangup
phoneA you will get a 200ok on it's leg.
That is why I asked you for the traces because it's most likely the case
t
I would recommend you join the IRC channel for more help... More
people can help you.
It's on irc.freeswitch.org in #freeswitch everyone in there is rather
friendly and we are all there to help you.
/b
On Apr 8, 2008, at 1:15 PM, kokoska rokoska wrote:
>
> Thank you, Antohny, for your reply
Anthony Minessale napsal(a):
> yes do you have both a pcap and a console trace of the call.
>
> start freeswitch with TPORT_LOG=1
>
> TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch
>
> set debug level
>
> > console loglevel debug
>
>
> capture all the text on the console
>
I have just u
Also, for now i think the 4th time, may i suggest you join our irc channel
so others may help me in assisting you?
I have been working fairly hard to keep up with your constant stream of
nearly realtime email requests.
On Tue, Apr 8, 2008 at 10:57 AM, Anthony Minessale <
[EMAIL PROTECTED]> wrote:
yes do you have both a pcap and a console trace of the call.
start freeswitch with TPORT_LOG=1
TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch
set debug level
> console loglevel debug
capture all the text on the console
my guess without seeing it is that user 22 is behind nat or a proxy or
Hi all,
I'm afraid I bug in FreeSWITCH SIP call handling.
The scenario is as follows:
Very simple dialplan:
User status:
Users 21 and 22 are registered, user 22 not.
Call flow:
User 21 calls number 23, recieve
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