Thanks for the assistance. I'm running version 1.0.6.7 of the software, tftp updated a few weeks ago.
I'm interfacing with an Asterisk box on my local lan. My sip.conf is as follows: [100] type=friend context=home callerid=Jim <100> secret=<mysecret> host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 mailbox=100 disallow=all allow=ulaw allow=gsm Can you recommend a method to which I can post the configuration from the grandstream bt100 device? Jim "Tony Mountifield" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > In article <[EMAIL PROTECTED]>, Jim Duda <[EMAIL PROTECTED]> > wrote: >> I knew about that one. I have Silence Suppression set to NO. > > Ah, ok. Puzzling then. If you'd like to post the full budgetone config > page(s), one of us might be able to spot something. > > What revision of budgetone firmware are you using? > > Is the budgetone talking to an Asterisk box of yours, or directly to > an external provider? > > Cheers > Tony > >> Jim >> >> >> "Tony Mountifield" <[EMAIL PROTECTED]> wrote in message >> news:[EMAIL PROTECTED] >> > In article <[EMAIL PROTECTED]>, >> > Jim Duda <[EMAIL PROTECTED]> wrote: >> >> -=-=-=-=-=- >> >> -=-=-=-=-=- >> >> >> >> I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old >> >> analog >> >> phones. The analog phones with the Sipura seem to work great. Voice >> >> quality is fine on both ends on the Sipura. I'm using the Teliax >> >> service >> >> and I use the Ulaw codec for all phones. >> >> >> >> However, I'm struggling with the BudgeTone 100. On my end, I find >> >> there >> >> is >> >> lot's of voice cut outs. I'm told my voice is find on the other end, >> >> but >> >> my >> >> receiving end gets the cutouts. I find it rather annoying and tend to >> >> always use the Sipura phones, which work great. >> >> >> >> I believe it's a configuration issue on the BudgeTone. I've followed >> >> all >> >> the examples and notes I could find on the subject on voip-info.com. >> >> >> >> Has anyone else had this experience with the BudgeTone? In general, I >> >> like >> >> the phone, wish it worked better. >> > >> > Turn OFF Silence Suppression in the Budgetone configuration. >> > >> > If SS is enabled, the phone stops sending RTP when you are silent. >> > Asterisk relies on the incoming RTP stream being continuous, using it >> > to generate the timing for the outgoing RTP. >> > >> > Cheers >> > Tony >> > -- >> > Tony Mountifield >> > Work: [EMAIL PROTECTED] - http://www.softins.co.uk >> > Play: [EMAIL PROTECTED] - http://tony.mountifield.org >> > _______________________________________________ >> > Asterisk-Users mailing list >> > Asterisk-Users@lists.digium.com >> > 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 >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > -- > Tony Mountifield > Work: [EMAIL PROTECTED] - http://www.softins.co.uk > Play: [EMAIL PROTECTED] - http://tony.mountifield.org > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users