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

Anthony Minessale anthony.minessale at gmail.com
Mon Jul 30 17:22:09 MSD 2012


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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