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" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, October 24, 2003 1:49 PM Subject: Re: [Asterisk-Users] Tonight's CVS breaks Grandstream phone
JT>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
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