<p>Thats understandable. Im cool with it, afterall its working. :)</p>
<div class="gmail_quote">On Jul 31, 2012 9:40 AM, "SamyGo" <<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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. <div>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.</div>
<div><br></div><div>Thanks a lot for your time on this.<br><div><br></div><div>Regards,</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">This script is actually a bit over-complicated<br>
<br>
dialA = "sofia/gateway/fs1/9903"<br>
dialB = "user/1001"<br>
legA = freeswitch.Session(dialA)<br>
<br>
if (legA:ready()) {<br>
legA:execute("bridge", dialB)<br>
}<br>
<br>
I think the problem is that we don't support dealing with many codecs<br>
in the 200ok<br>
They send gsm and ulaw and then choose ulaw and we go with the first one gsm<br>
<div><div><br>
<br>
<br>
On Sat, Jul 28, 2012 at 11:23 AM, SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>> wrote:<br>
> Hi again,<br>
><br>
> So, It was a very minor change in configuration and it was working.<br>
> Basically FreeSwicth was bridging the two legs BUT there was a codec issue.<br>
> All I had to do was in Asterisk (serving as my gateway) to allow only ulaw<br>
> and alaw<br>
> i.e<br>
><br>
> disallow=all<br>
> allow=ulaw<br>
> allow=alaw<br>
><br>
> What I really really wish to know is that why there was no indication of<br>
> codec mismatch or ptime mismatch or sample rate mismatch while transcoding<br>
> or anything.<br>
><br>
> It will be fine if none replies but it will be great to know the real<br>
> reason behind this and from where in logs can I verify this !!<br>
><br>
> Thanks<br>
> Sammy<br>
><br>
><br>
> On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>> wrote:<br>
>><br>
>> Here are the FS console logs:<br>
>> <a href="http://pastebin.freeswitch.org/19595" target="_blank">http://pastebin.freeswitch.org/19595</a><br>
>><br>
>> Please suggest what am I missing here.<br>
>><br>
>><br>
>> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>> wrote:<br>
>>><br>
>>> Hello,<br>
>>> I wanted to make a lua script which just dials out two different numbers<br>
>>> via some external gateway and when both calls are answered they are just<br>
>>> bridged. For this a very impressive Lua example<br>
>>> <a href="http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control" target="_blank">http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control</a> is copied and<br>
>>> all I had to do was change the dialA and dialB strings and its working great<br>
>>> as far as the SIP signalling is concerned.<br>
>>><br>
>>> execute this string and I get calls on two different number but things<br>
>>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not even<br>
>>> one-way. I could see on my own gateway that RTPs for both the legs are<br>
>>> actually forwarded to Freeswitch !<br>
>>><br>
>>> On my sip pcap traces analyzing on wireshark I could actually hear the<br>
>>> two persons saying Hello but neither could hear anything.<br>
>>><br>
>>> The above example lua call_control script is used as it is.<br>
>>> Please suggest.<br>
>>><br>
>>> Regards<br>
>>> Sammy Go.<br>
>>><br>
>><br>
><br>
><br>
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</blockquote></div><br></div></div>
</blockquote></div>