[asterisk-users] asterisk's SDP

[vip killa]

I have a call trace of one of these calls...and this seems strange:
asterisk sends on INVITE
a=fmtp:101 0-16
then 183 Session progress is sent back with:
a=fmtp:101 0-16
then asterisk sends 183 Session progress with:
a=fmtp:127 0-16
OK is sent back with:
a=fmtp:101 0-16
then asterisk sends OK with:
a=fmtp:127 0-16

Would the above cause DTMF not to be read on remote end?


On Fri, Jul 22, 2011 at 8:12 AM, vip killa <vipkilla at gmail.com> wrote:

> I see, thank you for explaning. The reason for my concern is, we are
> sometimes having DTMF issues on outbound calls. It seems when the user
> (Polycom) enters digits, they are not being recognized by the other end.
>
>
> On Thu, Jul 21, 2011 at 5:17 PM, Kevin P. Fleming <kpfleming at digium.com>wrote:
>
>> On 07/21/2011 03:54 PM, vip killa wrote:
>>
>>> What if asterisk sends telephony events that are not in range of 0-15
>>> though?
>>>
>>
>> You are misunderstanding how SDP works; when an SDP offer or answer is
>> sent, that indicates what the sender is willing to *receive*, not what it is
>> going to send.
>>
>> If the Sonus device sent "fmtp:101 0-15" in its SDP, then Asterisk should
>> not send 'event 16' events to it. If it does, that's a bug, although
>> standard programming practices would mean that it wouldn't be harmful, it
>> would just be ignored by the Sonus device.
>>
>>
>> --
>> Kevin P. Fleming
>> Digium, Inc. | Director of Software Technologies
>> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:
>> kpfleming
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>> Check us out at www.digium.com & www.asterisk.org
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