exactly,<br><br>The random crashes started happening by themselves, the protocol on the other end has exploited it's ancestor code and we really don't feel like bothering to fix it.<br><br><br><div class="gmail_quote">
On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <span dir="ltr"><<a href="mailto:sos@sokhapkin.dyndns.org">sos@sokhapkin.dyndns.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I had random crashes on IAX outgoing calls in mod_iax (all calls went to the<br>
same provider). I gave up and now use asterisk as protocol converter.<br>
<div><div></div><div class="h5"><br>
On Friday 05 February 2010, Meftah Tayeb wrote:<br>
> hi,<br>
> iax2 is secure<br>
> but, is not a good idea to avoid rtp and pass all packet including audio<br>
> and signalisation troug the same port<br>
> and digium added some change to the IAX2 protocol so freeswitch is not<br>
> up to date<br>
> no one want to update the iax2 stack in fs<br>
> so fs mod_iax have bean removedfrom the trunk<br>
><br>
> Le 05/02/2010 17:03, David Knell a écrit :<br>
> > There's a fairly simple solution to IAX needs, which is to run Asterisk,<br>
> > probably on the same box, as a protocol converter - you just need to<br>
> > tell it to use a non-standard port in sip.conf so that it doesn't clash<br>
> > with FreeSWITCH.<br>
> ><br>
> > --Dave<br>
> ><br>
> >> the lib that we used to provide iax support is pretty much abandonware<br>
> >> (no longer updated) and newer iax implementations (like latest<br>
> >> asterisk) can cause it to crash. There are no license compatible iax<br>
> >> implementations that work, so.. mod_iax has been moved to the<br>
> >> unsupported column.<br>
> >><br>
> >><br>
> >> Default passwords -- that is a single var in vars.xml that controls<br>
> >> the passwords.<br>
> >><br>
> >><br>
> >> number ranges - up to you. The sample configs supplied are just that,<br>
> >> samples. I use a smaller range personally.<br>
> >><br>
> >> On Fri, Feb 5, 2010 at 7:53 AM, Matthew Law<<a href="mailto:matt@webcontracts.co.uk">matt@webcontracts.co.uk</a>><br>
> >> wrote:<br>
> >> Why is that? - a lot of web pages I have read claim that IAX<br>
> >> is more<br>
> >> secure and efficient. I have no problem with using SIP<br>
> >> whatsoever and it<br>
> >> certainly appears to be ubiquitous. I am a complete newcomer<br>
> >> to VoIP and<br>
> >> I am trying to do this as securely as possible since I want to<br>
> >> run<br>
> >> freeswitch on a Xen VPS which will be visible on the internet.<br>
> >><br>
> >> Anyway, since my first question, I have worked my way through<br>
> >> the wiki,<br>
> >> read a lot of example configs and made some sense of the<br>
> >> docs. I now have<br>
> >> a very basic config working (with SIP) on a local vmware image<br>
> >> using the<br>
> >> 'quick and dirty' Makefile method. I removed all of the<br>
> >> example configs<br>
> >> from the xml files (those default extensions and passwords<br>
> >> scared me) and<br>
> >> added just one for extension 1000, plus my dialplan and<br>
> >> inbound/outbound<br>
> >> settings.<br>
> >><br>
> >> One question: is there any reason not to use a smaller<br>
> >> extension number<br>
> >> range, like 200-210, for example?<br>
> >><br>
> >> Now to figure out how to get time based roaming working...<br>
> >><br>
> >><br>
> >> Thanks,<br>
> >><br>
> >> Matt.<br>
> >><br>
> >> On Fri, February 5, 2010 6:43 am, Michael Jerris wrote:<br>
> >> > iax2 support has been removed from FreeSWITCH in current<br>
> >><br>
> >> trunk and will<br>
> >><br>
> >> > not be in the 1.0.5 release.<br>
> >> ><br>
> >> ><br>
> >> ><br>
> >> > Mike<br>
> >><br>
> >> _______________________________________________<br>
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> >><br>
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> >><br>
> >><br>
> >><br>
> >><br>
> >> --<br>
> >> -Rupa<br>
> >><br>
> >> _______________________________________________<br>
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> ><br>
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><br>
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