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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Chris Orme > > Sent: Saturday, April 10, 2004 6:37 AM > > To: [EMAIL PROTECTED] > > 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? > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users