currently 1.4.5 on bsd has issues.
1 because zaptel is out of sync asterisk does not see and
include zaptel in
its build. 1.4.5 seems to have the iax issue. Asterisk 1.4.6
should be
released soon to fix the current issues.
I have svn of branches/1.4 and the sip and iax issues are
not there but the
zaptel issue still is.
I have worked with killfill from the asterisk-bsd group to
get the ports
updated but we are waitng to hear back from Gonzo about
zaptel..
asterisk addons also fails to build currently. we are
working to fix this and
get the ports updated give us a fe more days.
On Monday 25 June 2007 09:28:05 Hendrik Visage wrote:
> Hi there,
>
> FreeBSD 6.2
> Asterisk 1.4.5 (and 1.4.3 from ports)
>
> Sip phone - PBX(*) -IAX2-VROUTER(*)- SIP-Voip provider
> (SPA901 & SPA922 phones)
>
> We've see a situation where the IAX2 appears to
"loose"/drop the voice
> data to be sent to the
> SIP side of things. This happens "semi"
intermittently, but we can
> reliably regenerate it
> at >40 alaw calls (even on a dedicated 1G network)
and also with G729
> (but a tad more calls).
> It appears to happen using both trunking and
non-trunking modes.
>
> This happened with DONT_OPTIMIZE setting on or off,
but with it ON it
> doesn;t dump core.
> At least when it was dumping core, it appeared to have
been in the
> pthread_cancel
> calls.
>
> We've recompiled the PBX asterisk with no threading,
and the
> milliwat/etc. tests to the vrouter
> from the SIP phones ran clean (other than when we
pushed the bandwidth
> limits <grin>)
>
> This morning it was consistently the agent (on the SIP
Phones) who
> could hear the remote side complaining that the remote
side can't hear
> them anymore. After we've recompiled the VROUTER
Asterisk with
> non-threading, the calls stayed stable.
>
> What appears to happen is that somewhere in the
threading the IAX
> voice data is discarded or something on the way to the
SIP side.
>
> Anybody else anything like this?
> Any other work around for this issue/problem??
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