Mike Jagdis wrote:
John Coll wrote:
Dave
You were right
disallow=all
allow=ulaw
allow=alaw
gave me two-way voice! Whew! Thanks a million. I wonder if I really
should
have found that for myself ... I've added it to the voip-info.org wiki
OK lets see if the next step is a bit easier :)
thanks again all
john
Note that if you don't have canreinvite=no you probably also want to
disable gsm on the GS phones themselves (just change the 723 entry in
the list on the admin page to a repeat of a 711).
Initially * negotiates each leg and relays packets. So the disallow
and allow in *'s config works. If reinvite is enabled * then about
10 seconds later the two end points will bounce SIP INVITES between
each other and start sending packets direct. Since * isn't in on
this negotiation the fact that it is configured to filter gsm out
of the codec list is immaterial...
I don't know if gsm actually works between GS phones or not, but it
definitely doesn't to other stuff. They negotiate gsm fine but send
gsm data to the rtp port and the GS phone replies with icmp errors.
Non-gsm data is fine...
Added to
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+grandstream+budgetone
Thank you!
Guess most of this also applies to the Handytone.
/O
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