search for: media_encryption

Displaying 18 results from an estimated 18 matches for "media_encryption".

2016 Mar 03
3
RTP / NAT question ( pjsip )
...cipher=AES256-SHA method=tlsv1 ;===============EXTENSION 6001 [6000] type=endpoint context=internal disallow=all allow=ulaw auth=auth6000 aors=6000 direct_media=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=no force_rport=yes rtp_symmetric=no media_encryption=sdes [auth6000] type=auth auth_type=userpass password=6000 username=6000 [6000] type=aor qualify_frequency=30 max_contacts=1 remove_existing=yes ;===============EXTENSION 6001 [6001] type=endpoint context=internal disallow=all allow=ulaw auth=auth6001 aors=6001 direct_media=no rewrite_con...
2015 Jul 08
6
tls on asterisk 13
...0:5061 cert_file=/etc/asterisk/keys/asterisk.crt priv_key_file=/etc/asterisk/keys/asterisk.key method=tlsv1 [XXXX] type=endpoint context=XX-Xip disallow=all allow=ulaw allow=alaw transport=transport-tls direct_media=no force_rport=yes rtp_symmetric=yes mailboxes=XXXX at default auth=XXXX aors=XXXX media_encryption=sdes dtmfmode=rfc4733 regardss -- rickygm http://gnuforever.homelinux.com
2018 Feb 08
3
pjsip trunking configuration issue
...= 0.0.0.0:5061 cert_file=cert_file priv_key_file=key_file method=tlsv1 external_media_address=X.Y.Z.D external_signaling_address=X.Y.Z.D verify_client=no verify_server=no allow_reload=yes [twilio](!) type=endpoint transport=transport-tls context=from-twilio disallow=all allow=ulaw dtmf_mode=inband media_encryption=sdes rtp_symmetric=yes rewrite_contact=yes force_rport=yes canreinvite=no tlsdontverifyserver=yes [auth-out](!) type=auth auth_type=userpass [twilio] aors=twilio-aors [twilio-aors] type=aor contact=sips:trunkname.pstn.twilio.com:5061 ;tried with sip: also [twilio] type=identify endpoint=twilio...
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
...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 > > 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, >> >> I am looking for documentation support fo...
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
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
2020 Apr 19
1
how to make a bug report
...try_interval = 200 transport = transport-tls endpoint/cos_audio = 5 endpoint/cos_video = 4 remote_hosts = secure.sip.easybell.de:5061 aor/qualify_frequency = 30 outbound_auth/username = ... outbound_auth/password = ... endpoint/allow = !all,g722,alaw,ulaw endpoint/context = ingressEasybell endpoint/media_encryption = sdes registration/contact_user = extenHW In pjsip.conf is only the transport: [transport-tls] type=transport protocol=tls bind=192.168.3.50:5061 ca_list_file=/etc/pki/tls/certs/ca-bundle.crt cert_file=/etc/asterisk/cert/newc/mycert.pem priv_key_file=/etc/asterisk/cert/newc/mykey.pem After I...
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
...> >>> 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>...
2015 Sep 15
3
Asterisk 13 WebRTC Status report
...ransport protocol=wss ;udp,tcp,tls,ws,wss bind=0.0.0.0 ;===============ENDPOINT TEMPLATES [endpoint-basic](!) type=endpoint transport=transport-wss context=route_phones disallow=all allow=alaw allow=ulaw force_avp=yes use_avpf=yes ; Determines whether res_pjsip will use and enforce usage of media_encryption=dtls ; Determines whether res_pjsip will use and enforce dtls_verify=no ; Verify that the provided 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....
2015 Mar 03
1
Cannot configure PJSIP TLS
Hey guys,tried to make tls work with pjsip on asterisk 13.2.0 have compiled pjsip with ssl, added transport [tls] type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061 local_net=192.168.1.0/24 external_media_address=77.77.77.77 external_signaling_address=77.77.77.77 have configured Grandstream GXP1400
2020 May 30
1
PJSIP
...ther question... how to set a=rtcp-fb Does asterisk support this ? Any help is much appreciated. PJSIP config. [test01] type=aor contact=sip:34.221.174.202:15033 [test01] type = endpoint transport=transport-tls ice_support = yes allow=!all,ulaw,h264 sdp_session=xaccel rtcp_mux=yes aors=test01 media_encryption=sdes from_user=109643183 from_domain=xaccel.net outbound_proxy=sip:34.221.174.202:15033 direct_media=no direct_media_method=invite dtmf_mode=rfc4733 use_avpf=yes John Bittner CTO [xaccellogoemail] 380 US Highway 46, Suite 500 Totowa, NJ 07512 Phone: 201.806.2602 x2405 Fax: 201.806.2604 Cell:...
2020 Apr 18
2
how to make a bug report
Hi, how do I make a bug report? I filled in the form to make a report and https://issues.asterisk.org/jira/issues/?filter=-2 still shows no issues reported by me. If someone knows how to get asterisk to re-register when using pjsip after the registration shows as Rejected, like after the internet connection to the VOIP provider goes away (and comes back), please let me know. This bug makes
2017 May 30
3
Asterisk 14.3.1 > 14.4.1 upgrade pjsip nat broken?
...cert_file=XXX ;removed priv_key_file=XXX ;removed bind=0.0.0.0:5061 external_media_address=x.x.x.x ;public ip external_signaling_address=x.x.x.x ;public ip local_net=192.168.0.0/16 [endpoint-common](!) type=endpoint context=users disallow=all allow=g722,ulaw,h264 dtmf_mode=info [endpoint-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=ye...
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?
..._at=0 dtls_cipher= from_domain= dtls_rekey=0 dtls_fingerprint=SHA-256 direct_media_method=invite send_rpid=false pickup_group= sdp_session=Asterisk dtls_verify=No message_context= 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 ca...
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
...low is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors=1060 auth=1060 [1060] type=auth auth_type=userpass password=1060 username=1060 [1060] type=aor max_contacts=10 [204] .... http.conf: enabled=yes bindaddr=10.10.5.49 bindport=8088 CLI> pjsip show endpoints Endpoint: 1060...
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)
2015 Apr 29
2
PJSIP - sessions-timers support not working on 13.X
...al issue [105] type=aor max_contacts=1 remove_existing=yes [105] type=auth auth_type=userpass password=XXXXXXXX username=105 [105] type=endpoint disallow=all allow=ulaw allow=alaw context=video-test auth=105 aors=105 direct_media=no force_rport=yes rewrite_contact=yes transport=transport-udp-nat media_encryption=no ice_support=no timers_min_se=90 ; Minimum session timers expiration period (default:; "90") timers=required ; Session timers for SIP packets (default: "yes") timers_sess_expires=3600 ; Maximum session timer expiration period now get things working and i coul...
2015 Apr 29
0
PJSIP - sessions-timers support not working on 13.X
...h_type=userpass > password=XXXXXXXX > username=105 > > [105] > type=endpoint > disallow=all > allow=ulaw > allow=alaw > context=video-test > auth=105 > aors=105 > direct_media=no > force_rport=yes > rewrite_contact=yes > transport=transport-udp-nat > media_encryption=no > ice_support=no > timers_min_se=90 ; Minimum session timers expiration period (default:; "90") > timers=required ; Session timers for SIP packets (default: "yes") > timers_sess_expires=3600 ; Maximum session timer expiration period > > &gt...