[Freeswitch-users] CCS_RINGING too early

[Hynek Cihlar]

Yes, it is outboubd from FS to somewhere else. Here's the console output
http://pastebin.com/Awme44SZ.

Hynek


On Sat, Feb 23, 2013 at 8:17 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The data provided is not really relevant to FS.
> If you want a diagnosis of what is happening in FS you need to provide FS
> logs from the console.
>
> I am not even 100% sure which direction this is but I think you mean
> outbound from FS to somewhere else.
>
> console loglevel debug
> sofia global siptrace on
>
>
>
> On Sat, Feb 23, 2013 at 11:38 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> There are some use-cases.
>>
>> The main one would be that you might get FS to generate a ringback tone
>> if you're using ignore_early_media=true since otherwise the caller would
>> never hear any ringing.
>>
>> -Steve
>>
>>
>>
>> On 23 February 2013 17:06, Hynek Cihlar <hynek.cihlar at gmail.com> wrote:
>>
>>> Is there even a valid use case for Freeswitch to set a channel to
>>> RINGING before the actual ring is signaled by the far endpoint? What other
>>> evidence would be helpful to diagnose the issue?
>>>
>>> Hynek
>>>
>>>
>>> On Tue, Feb 12, 2013 at 12:03 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:
>>>
>>>> When originating a call the respective call channel's call state is set
>>>> to RINGING right after progress 100 is received.
>>>>
>>>> Here's the captured flow:
>>>>
>>>> |Time     | <src ip>                              |
>>>> |         |                   | <dst ip>          |
>>>> |2.848    |         INVITE SDP (g711A g711U GSM
>>>> telephone-eventRTP...e-101 CN)          |SIP From: "" <sip:endpoint removed
>>>> To:<sip:endpoint removed
>>>> |         |(5080)   ------------------>  (5060)   |
>>>> |2.848    |         407 Proxy Authentication Required          |SIP
>>>> Status
>>>> |         |(5080)   <------------------  (5060)   |
>>>> |2.849    |         ACK       |                   |SIP Request
>>>> |         |(5080)   ------------------>  (5060)   |
>>>> |2.849    |         INVITE SDP (g711A g711U GSM
>>>> telephone-eventRTP...e-101 CN)          |SIP From: "" <sip:endpoint removed
>>>> To:<sip:endpoint removed
>>>> |         |(5080)   ------------------>  (5060)   |
>>>> |2.850    |         100 Trying|                   |SIP Status
>>>> |         |(5080)   <------------------  (5060)   |
>>>> |13.444   |         180 Ringing                   |SIP Status
>>>> |         |(5080)   <------------------  (5060)   |
>>>> |13.445   |         183 Session Progress SDP (g711A g711U GSM
>>>> tele...ne-eventRTPType-101)          |SIP Status
>>>> |         |(5080)   <------------------  (5060)   |
>>>> |13.445   |         RTP (g711A)                   |RTP Num packets:230
>>>>  Duration:4.574s SSRC:0x1E777E26
>>>> |         |(26056)  <------------------  (19312)  |
>>>> |13.601   |         RTP (g711A)                   |RTP Num packets:220
>>>>  Duration:4.419s SSRC:0xA530C519
>>>> |         |(26056)  ------------------>  (19312)  |
>>>>
>>>> After the 100 Trying is received the switch executes
>>>> switch_channel_perform_set_running_state (switch_channel.c) and the channel
>>>> call state is set to RINGING and ESL event CHANNEL_CALLSTATE:RINGING is
>>>> generated.
>>>>
>>>> I would expect the channel call state to be set to RINGING only after
>>>> 180 Ringing is received from the far endpoint.
>>>>
>>>> Could anybody give me a hint what could be wrong or what steps to take
>>>> to figure out? I am already out of ideas.
>>>>
>>>> Thanks!
>>>> Hynek
>>>>
>>>
>>>
>>> _________________________________________________________________________
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>>>
>>> 
>>> 
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>>
>> _________________________________________________________________________
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>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>>
>>
>
>
> --
> Anthony Minessale II
>
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
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