search for: rtp_keepal

Hi Joshua > The "rtp_keepalive" option can be used to have the RTP stack send an > RTP packet out. Try that and see what happens. Once again 'bullseye' that fixed the problem. Thank you! Mit freundlichen Gr?ssen -Beno?t Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ___________________...

Hello List Asterisk 13.14.1 in use with pjsip stack. On the remote side is a SBC which performs some 'nat' detection. I suppose this means the SBC listens from where it is getting RTP data and then replies to that ip. As long as the asterisk is initiating the call this is fine, the asterisk start sending RTP to the media IP of the SBC and the SBC is sending media back. Now I want to do

.../remove_existing = yes aor/qualify_frequency = 60 aor/qualify_timeout = 2000 endpoint/ice_support = no endpoint/disallow = g723,slin,ilbc,lpc10,g729,speex,g726aal2,g722 endpoint/allow = ulaw,alaw,adpcm,gsm endpoint/direct_media = no endpoint/force_rport = yes endpoint/rewrite_contact = yes endpoint/rtp_keepalive = 30 endpoint/rtp_symmetric = yes endpoint/rtp_timeout = 60 endpoint/rtp_timeout_hold = 60 endpoint/send_pai = yes endpoint/send_rpid = yes endpoint/trust_id_inbound = yes endpoint/trust_id_outbound = yes endpoint/trust_connected_line = no endpoint/send_connected_line = no endpoint/context = tru...

Asterisk 13.8 Is CALLERID(all) supposed to wok for pjsip? When I do this: exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) same => n,Dial(PJSIP/phone123, 30) I expect the callerid to be as set, but is always seems to be "phone123", the name of the endpoint. Andrew -------------- next part -------------- An HTML attachment was scrubbed... URL:

...xt= mailboxes= named_pickup_group= record_on_feature=automixmon dtls_private_key= named_call_group= t38_udptl_maxdatagram=0 media_encryption_optimistic=false aors=DEADDEADBEEF rpid_immediate=false outbound_proxy= identify_by=username inband_progress=false rtp_symmetric=false transport=transport-udp rtp_keepalive=0 t38_udptl_ec=none fax_detect=false t38_udptl_nat=false allow_transfer=true tos_video=0 srtp_tag_32=false timers_min_se=90 call_group= sub_min_expiry=0 100rel=yes direct_media=true rtp_timeout_hold=0 g726_non_standard=false dtmf_mode=rfc4733 voicemail_extension= rtp_timeout=0 dtls_cert_file= me...

On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: >>> Just a guess (without knowing about your network), but are the two ends >>> points on public networks and visible to one another? If not the reinvite >>> may be passing an internal (nat'ed)