[Freeswitch-dev] Bridge to PSTN Number Not Using Call Timeout Value

[Anthony Minessale]

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 at 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 at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, February 14, 2011 10:35 AM
> To: freeswitch-dev at 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 at 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 at lists.freeswitch.org
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
>> Anthony Minessale
>> Sent: Monday, January 31, 2011 12:13 PM
>> To: freeswitch-dev at 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 at 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 at 192.168.72.79:5060
>>> bridge({presence_id=2001 at 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 at g1)
>>>
>>>
>>>
>>> Thanks,
>>>
>>> Jerry
>>>
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>>> v
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>>>
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
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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
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> googletalk:conf+888 at 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 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