[Freeswitch-users] CCS_RINGING too early

Sat Feb 23 23:16:18 MSK 2013

I think I see why, nobody seems to use those callstate events enough to
notice anything wrong with them,
Try latest HEAD.

On Sat, Feb 23, 2013 at 2:02 PM, Hynek Cihlar <hynek.cihlar at gmail.com>wrote:

> 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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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
>>>
>>> _________________________________________________________________________
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>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
>> AIM: anthm
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>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-users mailing list
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>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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AIM: anthm
MSN:anthony_minessale at hotmail.com
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