search for: send_pai

...? 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_transport=no trust_id_inbound=yes s...

...unk between AST-A and AST-B is configured like this in pjsip.conf and is identical on both machines: [transport-lan] type=transport protocol=udp bind=0.0.0.0 tos=af31 [pbxbeta] type=endpoint disallow=all allow=g722 allow=ulaw transport=transport-lan context=phone-level3 aors=pbxbeta send_rpid=no send_pai=yes trust_id_inbound=yes trust_id_outbound=yes direct_media=yes direct_media_glare_mitigation=outgoing ;direct_media_method=update tos_audio=46 tos_video=34 t38_udptl=no t38_udptl_nat=no [pbxbeta] type=aor contact=sip:{remote IP address}:5060 [pbxbeta] type=identify endpoint=pbxbeta match={remote...

...s=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/asterisk-users/attachments/20160704/b9235a94/attachmen...

...sm,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 >> >...

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

...allow=alaw ; from phonetype (G.729 replaced with alaw) direct_media=no context=phone rtp_timeout=120 set_var=__phoneid=12 set_var=__contacttypeid=4 set_var=__phonelineid=78 callerid="Steve Murphy" <101> call_group=2 pickup_group=2 mailboxes=101 at murftest language=en send_rpid=yes send_pai=yes ?OK, that completes the config (I hope). Now, when I run "pjsip show endpoints, I get:? SFO02-HostedPBXPJSip-Dev03*CLI> pjsip show endpoints Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName....

...sallow = 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_hold = 60 endpoint/send_pai = yes endpoint/send_rpid = yes endpoint/trust_id_inbound = yes endpoint/trust_id_outbound = yes endpoint/trust_connected_line = no endpoint/send_connected_line = no endpoint/context = trunkhandler_pbx-sip-t1 Attached sip sessions and debug log... the only thing I found interesting was finding a...

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)

...point/allow = !all,g729 aor/qualify_frequency = 30 registration/expiration = 1800 contact_pattern=xxx [xxx](trunk_defaults) sends_auth = yes sends_registrations = yes endpoint/context = extensions remote_hosts = xxx.xx.xx.xx accepts_registrations = no endpoint/send_rpid = yes endpoint/send_pai = yes outbound_auth/username = xxx outbound_auth/password = xxx contact_pattern=xxx -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151019/1dcc4b09/attachment.html>

I'm being told by my ITSP that my Asterisk shouldn't be challenging their system to authenticate (i.e. a 401 response) when they send me a SIP MESSAGE (or I suppose a SIP INVITE for that matter). But I'm not sure what a pjsip.conf configuration for that looks like. How does one associate an incoming call/message with an existing authenticated outgoing registration so that Asterisk

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:

...at customer IP address. When caller id is present, the From must be the caller id number at ipitimi ip address and caller id is DID at customer IP address. The P-Asserted-Identity must be the DID at ipitimi ip address. For the endpoint, I have... from_domain = ipitimi ip address from_user = DID send_pai = yes If no caller id is present, calls go through IPitimi to my cell phone. However, if caller id is present, the P-Asserted-Identity is the caller id. Based on conversations with IPitimi and some other SIP products, this is incorrect. The P-Asserted-Identity should be the from_user at from_do...