[asterisk-users] asterisk's SDP

vip killa vipkilla at gmail.com
Fri Jul 22 08:17:41 CDT 2011


How can we wireshark a trace on the remote end? It is a peer such as Level3
or Dash

On Fri, Jul 22, 2011 at 9:15 AM, Jesie Paluca <jesie.paluca at gmail.com>wrote:

> Most likely if DTMF is not recognized on the far end, it would be an
> incompatibility setting of DTMF support or bug on either UAC and UAS.
>
> Wireshark trace at both end will help you understand the issue.
>
>
> On Fri, Jul 22, 2011 at 8:12 PM, 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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>>
>>
>> --
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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