[asterisk-users] RTP audio

[Jerry Geis]

On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis <jerry.geis at gmail.com> wrote:

> Has there been issues where "once in a while" RTP audio does not work ?
>
> Example: connection to Cisco call manager - works mostly all the time.
>
> once in a great while - person does not hear the "beep" when calling in.
> once in a great while - person they hear the beep - but do not hear the
> audio public address.
>
> What would I be looking for to track this beast down ?
>
> This is my SIP trunk
> [LSVOIP]
> type=friend
> dtmfmode=rfc2833
> secret=password
> username=LSVOIP
> defaultuser=LSVOIP
> disallow=all
> allow=ulaw
> allow=alaw
> context=incoming
> host=172.1.1.1
> canreinvite=yes
> qualify=yes
> insecure=invite
>
> Thoughts?
>
> Jerry
>


Is there any kind of pjsip vs old SIP (which I am using) issue happening
here. (asterisk 18.14.0)

Jerry
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