[Freeswitch-dev] mod_loopback no ringback tone

Anthony Minessale anthony.minessale at gmail.com
Tue Oct 26 07:21:54 PDT 2010


do you have inbound-bypass-media on?
or are you not setting {ignore_early_media=true} when you call loopback

The default config ext 9181 demonstrates this functionality and is
confirmed to work.



On Tue, Oct 26, 2010 at 3:09 AM, Bernhard Suttner
<bernhard.suttner at winet.ch> wrote:
> Hi,
>
> it does not work. I think the switch_core_get_partner function does not return the required session - so, the BOND_VARIABLE for sofia-/loopback-a is not set.
>
> Best regards,
> Bernhard
>
> -----Ursprüngliche Nachricht-----
> Von: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
> Gesendet: Montag, 25. Oktober 2010 21:22
> An: freeswitch-dev at lists.freeswitch.org
> Betreff: Re: [Freeswitch-dev] mod_loopback no ringback tone
>
> We try. We're not always on top of things as much as we'd like but we
> sure try =p
>
>
> On Mon, Oct 25, 2010 at 1:41 PM, Bernhard Suttner
> <bernhard.suttner at winet.ch> wrote:
>> Hi,
>>
>> thanks for your fast answer. I tried the version about 4 hours ago. I will re-check it tomorrow.
>>
>> Thanks a lot for your VERY FAST RESPONSES. Within the asterisk project it does need more than X days to get a response if the whole asterisk does crash :-)
>>
>> Best regards,
>> Bernhard
>>
>>
>> ----- Original Message -----
>> From: Anthony Minessale [mailto:anthony.minessale at gmail.com]
>> To: freeswitch-dev at lists.freeswitch.org
>> Sent: Mon, 25 Oct 2010 19:00:57 +0200
>> Subject: Re: [Freeswitch-dev] mod_loopback no ringback tone
>>
>>
>>> try again on latest HEAD that may have been broken over the weekend.
>>>
>>>
>>> On Mon, Oct 25, 2010 at 11:11 AM, Bernhard Suttner
>>> <bernhard.suttner at winet.ch> wrote:
>>> > Hi,
>>> >
>>> > I have the following scenario
>>> >
>>> > A ----(SIP) ---> FS ---(loopback)---> FS ----(SIP)---> B
>>> >
>>> > FS is of course the same Freeswitch Sever. If B sends RINGING to FS the
>>> RINGING will not be passed through to A because of the loopback module.
>>> (That is was the trace does say). What has to happen on mod_loopback if it
>>> does receive the RINGING? Does it have to do something like
>>> switch_channel_mark_ring_ready, switch_channel_ring_ready or something else?
>>> I think it has maybe something to do with the missing INDICATE_RINGING
>>> section in mod_loopback.
>>> >
>>> > Best regards,
>>> > Bernhard Suttner
>>> >
>>> > _______________________________________________
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>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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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
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>>>
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>>> pstn:+19193869900
>>>
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>>
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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
>
> _______________________________________________
> FreeSWITCH-dev mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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>
>
>
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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



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