[Freeswitch-users] Simple IAX2 setup - help with converting from asterisk to freeswitch
Anthony Minessale
anthony.minessale at gmail.com
Fri Feb 5 13:39:51 PST 2010
exactly,
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.
On Fri, Feb 5, 2010 at 3:32 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:
> 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 (no longer updated) and newer iax implementations (like latest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
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >>
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> > >>
> > >>
> > >>
> > >>
> > >> --
> > >> -Rupa
> > >>
> > >> _______________________________________________
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>
>
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--
Anthony Minessale II
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