[Freeswitch-users] One more thing..

[Doc]

I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet 
Size) from 0.030 to 0.020, and the error went away.  I was able to then 
test (and succeed) dialing from secondary skype user to skypiax, and 
have the call bridged automatically to one of my Sipura SPA-1001's.  I 
could speak into the phone, and hear it come through skype (on my 
laptop) no problem.  When I would talk on the laptop, I would get garble 
back on the phone, but that could be a sound issue on my laptop (my 
skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).

The interesting thing it did do, however, was eventually (within a 
minute?) It threw a couple of errors :

2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1
2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 
16619M[(nil)|37     ][ERRORA  826  ][interface1][-1, 5,21] EXIT? sent=-1

And on a subsequent test, I received no audio (the above error rolling 
on the console), and within a matter of seconds, FS crashed with this :

Segmentation fault (core dumped)

I have another request open on the group actually for the above error, 
so I think it'd make sense to discuss further there (rather than 
duplicating material).

Thanks!  At any rate, the error for the missing codec, etc, seemed to be 
gone once I updated the RTP time in my ATA.  Hope this helps someone!!

Brian West wrote:
> change the rtp time to 0.020 from 0.030
>
> /b