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@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the lib that we used to provide iax support is pretty much abandonware (no longer updated) and newer iax implementations (like latest asterisk) can cause it to crash.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@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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900