search for: rtp_timeout

Yes, this means the remote end was not sending any audio stream. But it shouldn't. According to [1] before remote end send OK or ACK there is one way SDP, no any audio stream. PJSIP channel (option rtp_timeout) does not take this one. Isn't it a mistake? What could be workarounds? 19.12.2016 11:33, Jean Aunis ?????: > > This means the remote end was not sending any audio stream, or the > audio stream was not received by Asterisk. The problem may have many > different reasons, but ofte...

Today I faced a problem. Please help to solve this problem. Asterisk 13.7 (chan_pjsip) & Zyxel Keenetic Plus DECT, firmware v2.06(AAGJ.9)C1 Outbound call from Zyxel Keenetic (pjsip endpoint) to PSTN (pjsip trunk). Call using early media (183 Session in progress) and rtp_timeout=10. After 10 seconds: [2016-12-16 13:53:15] NOTICE[6654] res_pjsip_sdp_rtp.c: Disconnecting channel 'PJSIP/xxx-0000027b' for lack of RTP activity in 10 seconds SIP dump is attached. According to [1] before called user agent send OK or ACK there is one way SDP. In sip dump (attached) i...

...sport-udp aors=murftest12 moh_suggest=default force_rport=yes rewrite_contact=yes rtp_symmetric=yes dtmf_mode=rfc4733 disallow=all allow=ulaw ; from phonetype allow=g722 ; from phonetype allow=alaw ; from phonetype allow=alaw ; from phonetype (G.729 replaced with alaw) direct_media=no context=phone rtp_timeout=120 set_var=__phoneid=12 set_var=__contacttypeid=4 set_var=__phonelineid=78 callerid="Steve Murphy" <101> call_group=2 pickup_group=2 mailboxes=101 at murftest language=en send_rpid=yes send_pai=yes ?OK, that completes the config (I hope). Now, when I run "pjsip show endpoint...

...lify_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 = trunkhandler_pbx-sip-t1 Attached sip sessions and debug log...

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:

...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= media_encryption=no media_use_received_transport=false direct_media_glare_mitigation=none trust_id_inbound=false force_avp=false record_off_feature=automixmon send_diversion=true language= mwi_from_u...

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)