keep updating, the maintainer of mod_skypeiax is adding new patches every few hours.
You can also join irc on irc.freenode.net #freeswitch and #freeswitch-dev to interact with him live.


One hint around here when we ask if you updated we work in 1 minute increments.



On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre@meanie.us> wrote:
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




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Anthony Minessale II

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