[Freeswitch-users] Simple IAX2 setup - help with converting from asterisk to freeswitch

[Sergey Okhapkin]

I had random crashes on IAX outgoing calls in mod_iax (all calls went to the 
same provider). I gave up and now use asterisk as protocol converter.

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 (no longer updated) and newer iax implementations (like latest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