[Freeswitch-users] LUA session:Bridge not actually bridging calls ~

Anthony Minessale anthony.minessale at gmail.com
Tue Jul 31 16:18:41 MSD 2012


Not at the moment.  If everyone is going to start doing this we may have to
change something.
On Jul 31, 2012 12:41 AM, "SamyGo" <govoiper at gmail.com> wrote:

> Though the problem is solved but is there any corresponding WARN or
> anything in the logs so that anyone can pick the problem right away.
> This fixing of codecs took alot of time and I was never expecting that
> this could be the cause Thinking that its Bridge() which is not doing its
> part.
>
> Thanks a lot for your time on this.
>
> Regards,
> Sammy
>
>
> On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> This script is actually a bit over-complicated
>>
>>  dialA = "sofia/gateway/fs1/9903"
>>  dialB = "user/1001"
>>  legA = freeswitch.Session(dialA)
>>
>> if (legA:ready()) {
>>  legA:execute("bridge", dialB)
>> }
>>
>> I think the problem is that we don't support dealing with many codecs
>> in the 200ok
>> They send gsm and ulaw and then choose ulaw and we go with the first one
>> gsm
>>
>>
>>
>> On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <govoiper at gmail.com> wrote:
>> > Hi again,
>> >
>> > So, It was a very minor change in configuration and it was working.
>> > Basically FreeSwicth was bridging the two legs BUT there was a codec
>> issue.
>> > All I had to do was in Asterisk (serving as my gateway) to allow only
>> ulaw
>> > and alaw
>> > i.e
>> >
>> > disallow=all
>> > allow=ulaw
>> > allow=alaw
>> >
>> > What I really really wish to know is that why there was no indication of
>> > codec mismatch or ptime mismatch or sample rate mismatch while
>> transcoding
>> > or anything.
>> >
>> > It will be fine if  none replies but it will be great to know the real
>> > reason behind this and from where in logs can I verify this !!
>> >
>> > Thanks
>> > Sammy
>> >
>> >
>> > On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper at gmail.com> wrote:
>> >>
>> >> Here are the FS console logs:
>> >> http://pastebin.freeswitch.org/19595
>> >>
>> >> Please suggest what am I missing here.
>> >>
>> >>
>> >> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper at gmail.com> wrote:
>> >>>
>> >>> Hello,
>> >>> I wanted to make a lua script which just dials out two different
>> numbers
>> >>> via some external gateway and when both calls are answered they are
>> just
>> >>> bridged. For this a very impressive Lua example
>> >>> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is
>> copied and
>> >>> all I had to do was change the dialA and dialB strings and its
>> working great
>> >>> as far as the SIP signalling is concerned.
>> >>>
>> >>> execute this string and I get calls on two different number but things
>> >>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not
>> even
>> >>> one-way. I could see on my own gateway  that RTPs for both the legs
>> are
>> >>> actually forwarded to Freeswitch !
>> >>>
>> >>> On my sip pcap traces analyzing on wireshark I could actually hear the
>> >>> two persons saying Hello but neither could hear anything.
>> >>>
>> >>> The above example lua call_control script is used as it is.
>> >>> Please suggest.
>> >>>
>> >>> Regards
>> >>> Sammy Go.
>> >>>
>> >>
>> >
>> >
>> >
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>>
>>
>> --
>> Anthony Minessale II
>>
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>>
>> 
>> 
>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
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