search for: media_use_received_transport

...wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no media_use_received_transport=no trust_id_inbound=yes send_pai=yes rtp_symmetric=yes rewrite_contact=yes message_context=astsms [200] type=endpoint aors=200 auth=200-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <200> dtmf_mode=rfc4733 use_avpf=no ice_support=no media_use_received_tr...

...gt; >> aors=100 >> >> auth=100-auth >> >> allow=ulaw,alaw,gsm,g726 >> >> context=from-internal >> >> callerid=device <100> >> >> dtmf_mode=rfc4733 >> >> use_avpf=no >> >> ice_support=no >> >> media_use_received_transport=no >> >> trust_id_inbound=yes >> >> send_pai=yes >> >> rtp_symmetric=yes >> >> rewrite_contact=yes >> >> message_context=astsms >> >> >> [200] >> >> type=endpoint >> >> aors=200 >> >> a...

...n the example > below : > > [100] > > type=endpoint > > aors=100 > > auth=100-auth > > allow=ulaw,alaw,gsm,g726 > > context=from-internal > > callerid=device <100> > > dtmf_mode=rfc4733 > > use_avpf=no > > ice_support=no > > media_use_received_transport=no > > trust_id_inbound=yes > > media_encryption=no > > rtp_symmetric=yes > > rewrite_contact=yes > > *message_context=astsms* > > > > On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello...

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

...t;>> >>> allow=ulaw,alaw,gsm,g726 >>> >>> context=from-internal >>> >>> callerid=device <100> >>> >>> dtmf_mode=rfc4733 >>> >>> use_avpf=no >>> >>> ice_support=no >>> >>> media_use_received_transport=no >>> >>> trust_id_inbound=yes >>> >>> media_encryption=no >>> >>> rtp_symmetric=yes >>> >>> rewrite_contact=yes >>> >>> *message_context=astsms* >>> >>> >>> >>> On Tue, No...

...vided peer certificate is valid (default: dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey dtls_cert_file=/etc/pki/tls/certs/pbx.crt dtls_private_key=/etc/pki/tls/private/pbx.key dtls_setup=actpass ice_support=yes ;This is specific to clients that support NAT traversal media_use_received_transport=yes [auth-userpass](!) type=auth auth_type=userpass [aor-single-reg](!) type=aor remove_existing=yes max_contacts=1 ;===============DEVICES [webrtc1](endpoint-basic) auth=webrtc1 aors=webrtc1 [webrtc1](auth-userpass) password=secret username=webrtc1 [webrtc1](aor-single-reg) relevant part o...

...point-sdes](!) media_encryption=sdes [aor-common](!) type=aor remove_existing=yes max_contacts=1 maximum_expiration=160 qualify_frequency=60 [207](endpoint-common,endpoint-sdes) ;Linphone callerid=Chris <PSTN number> auth=207 aors=207 mailboxes=201 at default use_avpf=yes rtp_symmetric=yes media_use_received_transport=yes force_rport=yes [207] type=auth auth_type=userpass password=supersecretpassword username=207 [207](aor-common) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170530/ee11e801/attachment.html>

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:

...=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_user= rtp_ipv6=false ice_support=false callerid=unknown aggregate_mwi=true one_touch_recording=false cos_video=0 accountcode= allow=(g722|ulaw|alaw) r...

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)