[Freeswitch-users] CCS_RINGING too early

[Anthony Minessale]

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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>>
>
> _________________________________________________________________________
> 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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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