[Asterisk-Users] Newbie - getting two local phonestocommunicate would be a good start :)

[Olle E. Johansson]

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