[Freeswitch-users] custom ringback behaviour

[Brian West]

You'll need to collect a sip trace.  I can't reproduce this issue so  
if you can do this:

TPORT_LOG=1 ./freeswitch

Then press F8 to turn on full debug log.... Collect that and put it on  
pastebin.freeswitch.org (user pastebin/ pass freeswitch)

Then reply with the link to the pastebin so we can see the sip trace.

/b

On Jun 16, 2008, at 4:01 AM, Cavalera Claudio Luigi wrote:

> freeswitch-users-bounces at lists.freeswitch.org wrote:
>> First of I recommend updating to SVN Trunk which will be 1.0.1 very
>> soon.
>
> Ok, now I've got a different problem.
> This first appeared before the upgrade, sadly I don't know what I have
> done to create this but I can't get rid of it anymore.
>
> Anyway, concerning my upgrade to SVN Trunk:
> when reinstalling even after a fresh svn checkout, configuration xml
> files do not get overwritten.
> I think this behaviour is good when you don't want to lose your  
> precious
> dialplan, but in my case I wanted to get rid of every possibile  
> cause of
> problems, therefore I did
> rm -rf /usr/local/freeswitch before make; make install; ...
> I hope this is enough to be sure to start from a perfect clean
> situation.
>
>
> Now when I issue my command through event socket:
> api originate sofia/default/12345 at switch
> &bridge(sofia/default/54321 at switch)
> I hear music on hold instead of ringback tone, moreover A and B can't
> speak because music on hold goes on even after B has answered.
>
> I think this is the relevant part of log debug:
> 2008-06-16 10:33:27 [DEBUG] switch_core_session.c:386
> switch_core_session_receive_message() Kill sofia/internal/12345 at switch
> [BREAK]
> 2008-06-16 10:33:27 [DEBUG] switch_core_session.c:386
> switch_core_session_receive_message() Kill sofia/internal/12345 at switch
> [BREAK]
> 2008-06-16 10:33:27 [DEBUG] switch_ivr.c:337 switch_ivr_parse_event()
> sofia/internal/12345 at switch Command Execute playback(local_stream:// 
> moh)
> 2008-06-16 10:33:27 [DEBUG] mod_local_stream.c:309
> local_stream_file_open() Opening Stream [moh/8000] 8000hz
> 2008-06-16 10:33:27 [DEBUG] switch_ivr_play_say.c:911
> switch_ivr_play_file() Codec Activated L16 at 8000hz 1 channels 20ms
>
> I have to solve this before going back to original problem :-)
>
>
> Thanks a lot,
> Claudio
>
>
>
>
>> On Jun 13, 2008, at 3:30 AM, Cavalera Claudio Luigi wrote:
>>> Now let's consider this couple of commands issued through fs console
>>> (I hope I'm using % vs @ properly here):
>>> 1) originate
>>> {ringback=/usr/local/freeswitch/sounds/calling.wav}sofia/default/
>>> 1000%my FsBox &bridge(sofia/default/0287654321 at MySswIP)
>>>
>>> in this case when I answer the first ringing phone (softphone) while
>>> the second phone is ringing I hear "calling.wav": wonderful!
>>>
>>
>> I personally don't use the %,  But it denotes to look locally for a
>> registered user vs trying to send the call to the remote domain.  In
>> both cases the ringback should be on the bridge side.
>>
>> originate sofia/default/1000%blah &bridge({ringback=blah.wav}sofia/
>> 1234 at switch)
>>
>> This seems to be one of the biggest things that people misunderstand.
>> Just because you set a variable on the A-Leg doesn't mean its
>> automatically on the B-Leg.  There are a few that are magically
>> exported to the B-Leg and I'll try to get those on the wiki to make
>> sure people know what they are.  The others you'll need to export or
>> set them on the B-Leg like I showed you in the example above.
>>
>>
>>
>>>
>>>
>>> 2)originate
>>> {ringback=/usr/local/freeswitch/sounds/calling.wav}sofia/default/
>>> 0212345 678 at MySswIP &bridge(sofia/default/0287654321 at MySswIP)
>>>
>>> in this case when I answer the first ringing phone I don't hear
>>> calling.wav, instead I hear normal ringback tone while the second
>>> phone is ringing.
>>>
>>> What could be the cause of these different behaviours?
>>>
>>> Thanks,
>>> Claudio
>>>
>>
>> Brian West
>> sip:brian at freeswitch.org
>>
>>
>>
>>
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Brian West
sip:brian at freeswitch.org