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

[John Todd]

>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