-MC

On Tue, Feb 15, 2011 at 12:57 PM, Jerry Richards <jerry.richards@teotech.com> wrote:

I tried setting ignore_early_media=true and ringback=${us-ring} but I still do not hear artificial ringback (see http://pastebin.freeswitch.org/15392). I also tried enable and disable of bypass-media but it didn't make any difference. Do you see what could be wrong?

Thanks,
Jerry

-----Original Message-----
From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale
Sent: Monday, February 14, 2011 1:51 PM
To: freeswitch-dev@lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

you are better off using ignore_early_media=true so the originates fail. When you allow early media, getting early media counts as success. When you are chaining many like that you can use artificial ringbak by setting the variable ringback to ${us-ring} on the A leg.

On Mon, Feb 14, 2011 at 3:44 PM, Jerry Richards <jerry.richards@teotech.com> wrote:
> Anthony,
>
> I have more than one XML bridge statement with continue_on_fail set true. In the case of early media and the bridge_answer_timeout, it looks like it does not continue and attempt the next bridge statement in the XML dialplan. True? If so, is there a way to make Freeswitch continue with the next bridge statement in this case?
>
> Thanks,
> Jerry
>
>
> -----Original Message-----
> From: freeswitch-dev-bounces@lists.freeswitch.org
> [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of
> Anthony Minessale
> Sent: Monday, February 14, 2011 10:35 AM
> To: freeswitch-dev@lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call
> Timeout Value
>
> set bridge_answer_timeout on the A leg
>
>
> On Mon, Feb 14, 2011 at 12:24 PM, Jerry Richards <jerry.richards@teotech.com> wrote:
>> I found out why my call_timeout was not kicking in when I call a PSTN number. It is because of early media. Is there a way to play early media and also have the call_timeout kick-in according to its setting?
>>
>> Thanks,
>> Jerry
>>
>> -----Original Message-----
>> From: freeswitch-dev-bounces@lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of
>> Anthony Minessale
>> Sent: Monday, January 31, 2011 12:13 PM
>> To: freeswitch-dev@lists.freeswitch.org
>> Subject: Re: [Freeswitch-dev] Bridge to PSTN Number Not Using Call
>> Timeout Value
>>
>> You should use {originate_timeout=N} but that may not be the actual problem.
>>
>> You would probably have to look at a trace with debug level etc to be sure.
>>
>>
>>
>> On Mon, Jan 31, 2011 at 2:07 PM, Jerry Richards <jerry.richards@teotech.com> wrote:
>>> Hello All,
>>>
>>>
>>>
>>> When I set the call_timeout variable (to 12 seconds) before a bridge
>>> to an outbound PSTN call, Freeswitch always rings for 24 seconds
>>> before going to voicemail. If I call an internal extension, then it
>>> uses the call_timeout variable. Do you know why it would do this?
>>>
>>>
>>>
>>> Here is the line that sets the call_timeout variable:
>>>
>>> switch_channel_set_variable(channel, "call_timeout", "6");
>>>
>>>
>>>
>>> Here is the PSTN bridge log:
>>>
>>> EXECUTE sofia/internal/2003@192.168.72.79:5060
>>> bridge({presence_id=2001@192.168.72.79}[lcr_carrier=Carrier /
>>> Location 1/INTERNAL PRI
>>> TRUNK,lcr_rate=0.00200,origination_caller_id_number=2003]openzap/smg
>>> _
>>> p
>>> rid/a/15248142341@g1)
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Jerry
>>>
>>> _______________________________________________
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>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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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@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888@conference.freeswitch.org
> googletalk:conf+888@conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
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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@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
googletalk:conf+888@conference.freeswitch.org
pstn:+19193869900

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