Re: RTP audio
Jerry Geis <[email protected]>
| Newsgroups | gmane.comp.telephony.pbx.asterisk.user |
|---|---|
| Message-ID | <CABr8-B7H+_hucCG+KxbzqZHRFMS=iYGo0OGTrPiz9q0uWRKGdQ@mail.gmail.com> |
On Tue, Oct 18, 2022 at 3:38 PM Jerry Geis <[email protected]> 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users