search for: disable_direct_media_on_nat

Displaying 6 results from an estimated 6 matches for "disable_direct_media_on_nat".

2020 May 17
1
PJSIP sending RTP to private address
My phone is located behind a NAT, 172.16.0.0/21. Asterisk 16 is on a public IP. PJSIP has the config below: force_rport=yes direct_media=yes disable_direct_media_on_nat = yes direct_media_method=invite But when I send a call I see the RTP being sent to my private address, vs the public IP. This only happens when Asterisk has dialed the call to another carrier. If instead of Dial I choose Answer() and MusicOnHold, then the RTP gets shipped to the right address. T...
2014 May 07
0
Video with asterisk12 and pjsip
...tting RTP source address to 192.168.8.203:31384 Endpoint 7000 is a Grandstream GXV3175 with Video the pjsip.conf for exten 7000 is [7000] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw,h264,h263p,h263,h261 transport=transport-udp auth=auth7000 aors=7000 direct_media=no disable_direct_media_on_nat=yes do I have to turn on the Video Support somewhere else ? -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/atta...
2015 Mar 04
1
PJSIP: Failed to create outgoing session to endpoint
...xec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination) What settings has mistake? What logic to choose outgoing transport? [transport-udp] type=transport bind=0.0.0.0:5070 protocol=udp [srv_d228] type=endpoint language=ru rtp_symmetric=yes force_rport=yes disable_direct_media_on_nat=yes rewrite_contact=yes ice_support=yes disallow=all allow=alaw context=ext-fromservers from_domain=sipnet.ru from_user=talk37.ru aors=srv_d228 auth=srv_d228 set_var=fromDeviceId=228 set_var=fromUserId=2 outbound_auth=srv_d228 ;outbound_proxy=sip:sipnet.ru:5060 transport=transport-udp [srv_d228]...
2016 Jul 01
2
CALLERID on pjsip doesn't work?
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:
2016 Jul 04
2
CALLERID on pjsip doesn't work?
...e callerid=unknown aggregate_mwi=true one_touch_recording=false cos_video=0 accountcode= allow=(g722|ulaw|alaw) rewrite_contact=false t38_udptl_ipv6=false tone_zone= user_eq_phone=false allow_subscribe=true rtp_engine=asterisk auth=DEADDEADBEEF from_user=DEADDEADBEEF bind_rtp_to_media_address=false disable_direct_media_on_nat=false set_var= use_ptime=false outbound_auth= media_address= tos_audio=0 dtls_ca_path= dtls_setup=active force_rport=false connected_line_method=invite callerid_tag= timers=yes sdp_owner=- trust_id_outbound=false use_avpf=false context=default moh_suggest=default send_pai=false t38_udptl=false dtls...
2017 Jun 05
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
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)