[Asterisk-Users] Re: BudgeTone 100 Woes

Jim Duda jim at duda.tzo.com
Sun Aug 7 12:59:47 MST 2005


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" <tony at softins.clara.co.uk> wrote in message 
news:dd5kpa$ajh$1 at softins.clara.co.uk...
> In article <dd3be4$e01$1 at sea.gmane.org>, Jim Duda <jim at duda.tzo.com> 
> 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" <tony at softins.clara.co.uk> wrote in message
>> news:dd34rd$5hr$1 at softins.clara.co.uk...
>> > In article <20050806145704.2A27917EA7 at linux.duda.tzo.com>,
>> > Jim Duda <jim at duda.tzo.com> 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: tony at softins.co.uk - http://www.softins.co.uk
>> > Play: tony at mountifield.org - http://tony.mountifield.org
>> > _______________________________________________
>> > Asterisk-Users mailing list
>> > Asterisk-Users at 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 at 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: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at 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
> 






More information about the asterisk-users mailing list