[Freeswitch-users] CCS_RINGING too early

Anthony Minessale anthony.minessale at gmail.com
Sun Feb 24 01:26:08 MSK 2013


Head is the most stable branch currently, its in final stage. We will not
merge things until 1.2 is branched and 1.5 is tagged on head.
On Feb 23, 2013 3:51 PM, "Hynek Cihlar" <hynek.cihlar at gmail.com> wrote:

> With the HEAD it now works as expected, RINGING comes only after 180/183
> is received from the originated endpoint.
>
> Could you also merge the change into more stable branch? Do you want me to
> file a bug for that?
>
> Hynek
>
>
> On Sat, Feb 23, 2013 at 9:16 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> 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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>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> 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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>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
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>>>> googletalk:conf+888 at conference.freeswitch.org
>>>> pstn:+19193869900
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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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>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> 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/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> 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
>>
>>
>
> _________________________________________________________________________
> 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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>
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