Hi all, I'm setting up my first * install and have it peering with another * machine using IAX across the internet which provides our pstn gateway. So far I have the IAX "friend" set up correctly but when I make a test call from an external phone, I get: WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to calculate timelen on 8 packets I have set up a test default context as follows: [default] exten => pstnnumber,1,Answer exten => pstnnumber,2,Playback,demo-thanks exten => pstnnumber,3,Hangup Which doesnt appear to be loading the Playback module Any ideas? Dave
Can you try iax2? mark On Thu, 14 Aug 2003, Dave Wilson wrote:> Hi all, > > I'm setting up my first * install and have it peering with another * machine > using IAX across the internet which provides our pstn gateway. > > So far I have the IAX "friend" set up correctly but when I make a test call > from an external phone, I get: > > WARNING[5126]: File chan_iax.c, Line 648 (get_timelen): Don't know how to > calculate timelen on 8 packets > > I have set up a test default context as follows: > > [default] > > exten => pstnnumber,1,Answer > exten => pstnnumber,2,Playback,demo-thanks > exten => pstnnumber,3,Hangup > > Which doesnt appear to be loading the Playback module > > Any ideas? > Dave > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
Hopefully these are all correct: -No, a SIP phones cannot connect to Asterisk voicemail using G.729 if you do not have a licence. You will need a licence for at least one channel to do this. -DTMF is not related to the codec itself or how it works, however, inband DTMF is not recommended on compressed channels due to the possibility of distortion. Outband is the way to go as the tones are capture and then regenerated locally. Inband signaling is achieved thought the telephony voice channel, while outband is through some other type of communication channel. In the case of SIP it is the INFO method. Hope this helps, J On Thu, 14 Aug 2003 09:03:15 -0700 "George Lin" <glin@cosini.com> wrote:>*This message was transferred with a trial version of >CommuniGate(tm) Pro* > >Dear all, > >I like to know if the DTMF option is related to the codec >or not. Can a SIP >phone with g729 codec to access asterisk voicemail2 in >case the asterisk >does not have g729 license ?? If yes, what is the DTMF >option inband or >outband ??? Is there any successful experience ??? > >Regards, > >George Lin > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-usersRegards, Jamie Carl Jazz Inc. Email: me@jazz-inc.net Web: www.jazz-inc.net Phone: +61-414-365-466 Jabber: jazz@netmindz.net