Dave I note your suggestion "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"
My GS phone has the following codec options: PCMU, PCMA, G.723.1, G729A/B, G726-32, G728. Half an hour's research and reading tells me that PCMU and PCMA are G.711. Can you confirm Dave, that I should ONLY have PCMU and PCMA in all the six options that GS provide for selecting codecs - or is it OK to have G.729A/B, G.726-32 and G.728 but not to have G.723.1? thanks john -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson Sent: 04 January 2004 17:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :) 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+budg etone Thank you! Guess most of this also applies to the Handytone. /O _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users