[Asterisk-Users] No ringing tone with IAXY (and other bits and bobs)

[Rich Adamson]

Brian,

I need to roll back to an earlier version to identify a different problem,
but I dont have the cvs checkout command string that includes a date. Can
you post how to do that please?

Rich

------------------------
> What version of the Asterisk code are you running? 1_0 stable is definitely
> broken wrt ringback, and the latest stuff seems really broken in all kinds
> of ways. After seeing that others were having similar problems, and that
> someone had solved many of them by rolling back to the CVS version from 3/5,
> I tried the same and things are working marvelously (well, mostly).
> 
> -brian 
> 
> > -----Original Message-----
> > From: asterisk-users-admin at lists.digium.com 
> > [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Chris Orme
> > Sent: Saturday, April 10, 2004 6:37 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] No ringing tone with IAXY (and 
> > other bits and bobs)
> > 
> > Hi!
> > 
> > I'm really hope you can help me solve a little mystery, the 
> > mystery is probably just my misunderstanding ! sorry...
> > 
> > I've got an iaxy talking to my * box which connects to two providers.
> > I'm running the stable release of the pbx.
> > 
> > The only thing is that when dialling from the iaxy the 
> > ringing tone isn't heard while calling someone - you just 
> > hear silence then, they either answer or they don't on the remote end.
> > 
> > >From my extensions.conf is the following - I tried putting the ,r in 
> > >and
> > it doesn't help.  Is there some other option I could try here ?
> > 
> > Also I'm getting quite a bit of echo noticed at the remote 
> > end as well as the iaxy end.  All lines are digital, I guess 
> > only the jitter buffer is there to be tweaked to try and help ?
> > 
> > There is also this echo problem with the sipura, but not with 
> > an ATA186 or snom.  The lack of a ringing tone is only with the iaxy.
> > 
> > The Answer,Hangup lines were to solve 'busy' situations with 
> > SIP phones, without this or even with 'Congestion' they just 
> > rang forever if a number was busy.  They seem to need the 
> > 'Answer' line.
> > 
> > If you know a nicer or more correct way for me to do this 
> > please let me know as most times the SIP phone user will hear 
> > half a ring and then the hangup noise generated by the SIP 
> > device when a number they call is busy.
> > 
> > Many thanks!!
> > 
> > Chris  
> > 
> > PS please Cc: me a copy as well as to the list in case I miss 
> > it - Thanks.
> > << extensions.conf >> 
> > 
> > exten => _00.,1,AbsoluteTimeout(3600)
> > exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r)
> > exten => _00.,3,Answer
> > exten => _00.,4,Hangup
> > exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r)
> > exten => _00.,104,Answer
> > exten => _00.,105,Hangup
> > 
> > <<iax.conf>>
> > 
> > [iaxy]
> > type=friend
> > accountcode=iaxy
> > disallow=all
> > ;;allow=adpcm
> > allow=ulaw
> > username=iaxy
> > secret=xxx
> > auth=md5
> > nat=yes         <- nat=1 ??
> > notransfer=yes  <-this doesn't seem to work, perhaps in the 
> > wrong order?
> > host=dynamic
> > qualify=10000
> > 
> > Is the definitive order these should be in listed anywhere as 
> > I know it really seems critical and lines can be ignored if 
> > they're not in spot on the right order?
> > 
> > _______________________________________________
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> 
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