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

[John Coll]

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: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Olle E.
Johansson
Sent: 04 January 2004 17:03
To: asterisk-users at lists.digium.com
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

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