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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