It's a good shout but sadly hasn't helped. Thanks anyway! The issue seems to be that our provider expects to be able to send inband early media. There is an OpenSIPS box between the provider & Asterisk which essentially just routes SIP traffic so the behaviour at our end is still controlled by Asterisk which makes the call.
Using dtmfmode=auto it seems to be possible to switch to inband if RFC2833 is not advertised in SDP but the provider just honours what we set in the call setup, which, since we only use RFC2833 is always advertised in SDP. ATM I think it's a provider issue, according to another environment they should never send us inband but it seems to not be working correctly in the case. Regards Mark. On Mon, 17 Jun 2019 at 10:11, Floimair Florian <f.floim...@commend.com> wrote: > Just a guess, but I suspect that this might be related with strictrtp > setting in rtp.conf, which learns the correct source in doing so drops a > few packets. > > I would try to disable strictrtp for testing purposes and if this works > add some delay before playing back the media. > > > > > > > > With best regards > > > > *Florian Floimair *Innovation - Software-Development > > > *COMMEND INTERNATIONAL GMBH *A-5020 Salzburg, Saalachstraße 51 > http://www.commend.com > > > > *Security and Communication by Commend *FN 178618z | LG Salzburg > > > > *Von: *asterisk-users <asterisk-users-boun...@lists.digium.com> im > Auftrag von Mark Farmer <farm...@gmail.com> > *Antworten an: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Datum: *Freitag, 14. Juni 2019 um 15:15 > *An: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Betreff: *[asterisk-users] Early Media Issue > > > > Hi all > > > > I've got an issue where when I call a number that just plays early media > back to me. > > Instead of hearing the full sequence of tones I hear a short ringing then > part of the sequence. What seems odd is that I can see > the telephone-event/8000 being passed up the chain but when it gets to > Asterisk, it is never sent back to the phone. Instead I just see the usual > RTP flows. > > > > I've been trying to fix this for hours, does anyone have any ideas how to > get this working correctly? > > > > Asterisk version is 13.25.0 > > > > The settings I think are relevant (I'm using chan_sip): > > > > (sip.conf) > > ignoresdpversion=yes > > internal_timing=yes > > progressinband=never > > silencesuppression=no > > prematuremedia=no > > > > (Per peer) > > progressinband=yes > > directrtpsetup=no > > dtmfmode=rfc2833 > > directmedia=no > > silencesuppression=no > > prematuremedia=no > > > > > > TIA > > Mark. > > > > -- > > Mark Farmer > farm...@gmail.com > -- > _____________________________________________________________________ > -- 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 -- Mark Farmer farm...@gmail.com
-- _____________________________________________________________________ -- 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