[Asterisk-Users] Tonight's CVS breaks Grandstream phone

[John Todd]

Did you have * working before the "latest" CVS updates of a few days ago?

Try:

disallow=all
allow=ulaw
allow=alaw


and see how that works for you.  Put those lines into any SIP entries 
in sip.conf to make doubly sure you've got all your permissions 
straight.  I have tested with my grandstream 102 and Asterisk 
CVS-10/24/03-01:48:29 and I get everything working OK between the GS 
phones, zap cards, and Cisco SIP phones.

JT


>Hi Guys,
>
>Tried the disallow=all and allow=all but still getting one way audio with
>x-lite and messenger.
>
>Any update on this problem.
>
>Dave
>----- Original Message -----
>From: "John Todd" <jtodd at loligo.com>
>To: <asterisk-users at lists.digium.com>
>Sent: Friday, October 24, 2003 1:49 PM
>Subject: Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone
>
>
>>  >FYI.  Haven't dug enough to be able to report any more, but
>>  >re-fetched CVS to verify that sometime in the last few days CVS
>>  >changes now break my GS phone.
>>  >
>>  >It appears to be at the RTP level.  It seems to set the call up just
>>  >fine, but no audio is passed back to the instrument.
>>  >
>>  >I reverted, and will try to play with this tomorrow unless someone
>>  >else tells us it's fixed.
>>  >
>>  >Thx.
>>  >
>>  >B.
>>
>>  I am seeing the same error with CVS as of 02:00 today GMT.
>>  Grandstream phones will dial, the dialplan will seem to work, but
>>  after a few seconds the call fails.  Looking at the SIP debug, I see
>>  that
>>
>>  There was a new feature added last night to allow for codec
>>  permission/denial on a per-peer basis in sip.conf.  This means that
>>  each SIP client can be forced to use particular codecs (at least,
>>  that is the intent.  more testing, anyone?)
>>
>>  So, it seems that the Grandstreams do not elegantly handle some
>>  circumstances of codec presentation which were created by these new
>>  patches.  It is necessary for you to put the following lines in each
>>  Grandstream entry in your sip.conf, OR you can put the identical
>>  entries in [general] to have it work across all clients.  Note that
>>  both lines are required:
>>
>>  disallow=all
>>  allow=all
>>
>>
>  JT