search for: use_avpf

...ome parameter 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 u...

..., >> >> [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 >> >> >> [...

...conf ? >> On a fresh 13.8.2, it doesn't seem but I may have missed somthing. >> > > I thought it did. Let me check. > It's working for me... [1000] type = wizard endpoint/set_var = xxxxx=ttttttttt > pjsip show endpoint 1000 ... trust_id_outbound : true use_avpf : false use_ptime : false user_eq_phone : false voicemail_extension : xxxxx : ttttttttt > > >> >> Regards >> >> -- >> _______________________________________________________...

...nt_custom.conf and add 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 <...

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

...; I thought it did. Let me check. >>> >> >> It's working for me... >> >> [1000] >> type = wizard >> endpoint/set_var = xxxxx=ttttttttt >> >> > pjsip show endpoint 1000 >> ... >> trust_id_outbound : true >> use_avpf : false >> use_ptime : false >> user_eq_phone : false >> voicemail_extension : >> xxxxx : ttttttttt >> >> >> > I was looking for a set_var=xxxxx setting while I should...

...t >>> >>> 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 >>> &...

...g support p.s.2 relevant part of pjsip.conf [global] [transport-wss] type=transport 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...

Hello, I've just discovered PJSIP 's support of set_var setting in pjsip.conf. Is this setting also supported in pjsip_wizard.conf ? On a fresh 13.8.2, it doesn't seem but I may have missed somthing. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

...[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: 973.390.1090 www.xaccel.net<http://www.xaccel.net/> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is for the sole use...

...,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=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-user...

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:

...ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (Reported by James Van Vleet) * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POOD...

...ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (Reported by James Van Vleet) * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections (Reported by Matt Jordan) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POOD...

...EEF 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_ca_file= callerid_privacy=allowed_not_screened mwi_subscribe_replaces_unsolicited=false cos_audio=0 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/ast...

...direction? Below 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...

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)

...Corey Farrell) * ASTERISK-24435 - Asterisk 13 with TC400P segfault (Reported by Marian Koniuszko) * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566) (Reported by abelbeck) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (Reported by James Van Vleet) * ASTERISK-24381 - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections (Reported by Matt Jordan) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending quali...

...by Corey Farrell) - [ASTERISK-24435 <https://issues.asterisk.org/jira/browse/ASTERISK-24435>] - Asterisk 13 with TC400P segfault (Reported by Marian Koniuszko) - [ASTERISK-24122 <https://issues.asterisk.org/jira/browse/ASTERISK-24122>] - Documentaton for res_pjsip option use_avpf needs to be fixed (Reported by James Van Vleet) - [ASTERISK-24381 <https://issues.asterisk.org/jira/browse/ASTERISK-24381>] - res_pjsip_sdp_rtp: Declined media streams are interpreted, leading to erroneous 488 rejections (Reported by Matt Jordan) - [ASTERISK-24063 <https://i...