[Asterisk-Users] Asterisk SIP + Grandstream 100 phone

[WipeOut .]

My sip .conf entry for the gs phone looks like this..

[2010]
type = friend
context = local
sectet = xxx
host = dynamic
callerid = Name <2010>

Your sound issue sounds like a codec problem, make sure you have not disabled ulaw or alaw cos these are the only codecs that work with the GS phones.. Unless you have purchased G.729 licenses in which case you can use that as well..

THese phones are vey simple so you should be able to get it working easily..

Later..

> hi ..
> 
> i've just converted myself back to a newbie by trying to experiment with 
> some new stuff .. I have connected two grandstream Budgettone 100 phones 
> to my asterisk, and trying to experiment with them ..
> 
> I am trying to get into the asterisk sample basically ..
> when I dial 1000 asterisk receives the call, but I do not
> hear any sound on the phone.
> 
> Dialling from phone to phone direct (via IP) works OK so nothing is
> wrong with the phones..
> 
> Has someone had any experience with these telephones, and maybe
> can send me working entries for sip.conf etc. to save me all the
> messing about.
> 
> Also I cannot get my phone to register with asterisk, it keeps
> on giving a "registration failed". My sip.conf entry is:
> 
> [mvp]
> type=friend
> username=mvp
> fromuser=mvp
> secret=blah
> defaultip=192.168.0.210
> 
> on the phone I set authentication user mvp, and password blah ..
> 
> someone enlighten me please :) :)
> 
> cheers
> Dave
> 
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