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...
Mike
P.S. Asterisk <-> (say) X-Lite using gsm is fine...
-- Mike Jagdis Web: http://www.eris-associates.co.uk Eris Associates Limited Tel: +44 7780 608 368 Reading, England Fax: +44 118 926 6974 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
