search for: ice_support

...in common so i 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_...

.../dev1.key ca_list_file=/etc/asterisk/keys/ca.crt 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=ula...

Hello, Where can I find doc about PJSIP's ice_support parameter ? Do you need to configure things elsewhere in Asterisk config files (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? I'm asking because, if I'm not mistaken, STUN requires setting a STUN server so I think ICE most probably, should also require settings some...

...> >> 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=endpo...

...box > at time and when things go wrong (like this) it's hard to dig and figure > out what is up. > Here is more information from the browser about the session: https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the endpoint. I have configured a STUN server in both rtp.conf and res_stun_monitor.conf -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)8116-9161

...gnaling_address to the pjsipconfig registration seams to work, but knowing that the IP could change really means I need some kind of STUN/ICE similar to what we ran with chan_sip. I can find limited documentation on this, and what I have found does not show how to set a stun server to make the ice_support field work on an endpoint. Can anyone advise where I could find an answer to this. Thanks in advance for any ideas you can offer. Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/201...

...the message context as in 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 a...

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

...gt;>> >>> 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* >...

...hether 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.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](...

...t                  = 10.10.41.0/24 local_net                  = 10.5.1.0/24 pjsip_wizard ------------- [isoft-sr-in-1] type = wizard transport = transport-udp remote_hosts = 192.81.237.20 aor/max_contacts = 1 aor/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_ho...

On Tue, Oct 27, 2020 at 5:35 AM Olivier <oza.4h07 at gmail.com> wrote: > Hello, > > Where can I find doc about PJSIP's ice_support parameter ? > > Do you need to configure things elsewhere in Asterisk config files > (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? > It needs to also be enabled in rtp.conf. > I'm asking because, if I'm not mistaken, STUN requires setting a STUN &gt...

...P' (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] type=aor qualify_frequency=30 contac...

...don't know how this server really works) Cheers Le mar. 27 oct. 2020 à 09:53, Joshua C. Colp <jcolp at sangoma.com> a écrit : > On Tue, Oct 27, 2020 at 5:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hello, >> >> Where can I find doc about PJSIP's ice_support parameter ? >> >> Do you need to configure things elsewhere in Asterisk config files >> (rtp.conf, PJSIP transport sections, ...) to make ICE work properly ? >> > > It needs to also be enabled in rtp.conf. > > >> I'm asking because, if I'm not mista...

...quot; <sip:5tf2f2s0rbtdj-20d14fl6n65t0o-0u03@ infront of the ip address. ? Another 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 Highwa...

...to the pjsipconfig registration seams to > work, but knowing that the IP could change really means I need some kind > of STUN/ICE similar to what we ran with chan_sip. > I can find limited documentation on this, and what I have found does not > show how to set a stun server to make the ice_support field work on an > endpoint. > Can anyone advise where I could find an answer to this. > Thanks in advance for any ideas you can offer. > Bryant The res_pjsip module does not currently support an auto-updating mechanism for the external signaling and media address information. -- Josh...

...with this PJSIP settings: --- /etc/asterisk/pjsip.conf --- [transport-udp] type=transport protocol=udp bind=0.0.0.0 [6001] type=endpoint transport=transport-udp context=from-internal disallow=all allow=ulaw auth=6001 aors=6001 direct_media=no rtp_symmetric=yes force_rport=yes rewrite_contact=yes ice_support=yes [6001] type=auth auth_type=userpass password=P at ssw0rd username=6001 [6001] type=aor max_contacts=2 [...] --- EOF --- And this is my dialplan: --- /etc/asterisk/extensions.conf --- [from-internal] exten => 100,1,Answer()    same => n,Wait(1)    same => n,Playback(hello-world)...

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:

...The chan_pjsip module doesn't prevent that. You'd need to provide the > full SUBSCRIBE now that it is actually finding the endpoint and coming > in. Ok, let's see if we can solve the mystery.. pjsip.conf [endpt-home](!) type=endpoint disallow=all allow=g722 allow=alaw allow=gsm ice_support=yes context=from-home allow_subscribe=yes mwi_subscribe_replaces_unsolicited=yes [11](endpt-home) auth=11 aors=11 callerid=*scrubbed* mailboxes=11 voicemail_extension=411 extensions.conf [from-home] exten => 11,hint,PJSIP/11 include=>local-extens include=>trunk-out This is the exact s...

...(Reported by Paulo Vicentini) * ASTERISK-26711 - func_enum: ENUM code wrong case (Reported by Vitold) * ASTERISK-23407 - Fix the FSF address in the headers of lots of pjproject files (Reported by Jared Smith) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_support is enabled but not used (Reported by Torrey Searle) * ASTERISK-19460 - [patch] Function TXTCIDNAME never actually makes DNS calls and always returns an empty string (Reported by George Joseph) New Features made in this release: ----------------------------------- *...