search for: trust_id_outbound

...ting also supported in pjsip_wizard.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 >> >> -- >> ____________________________...

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

...figured 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 IP address} The phones have the followin...

...ng. >>>> >>> >>> 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 s...

...,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 lack of a log item We SHOULD be seeing: DEBUG[XXXXX] res_pjsip_session.c: (null session):...

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:

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:

...d [didassignedbybluip] type = endpoint context = TestApp transport = transport1 outbound_auth = auth7 aors = BLUIPIN accountcode = 16 dtmf_mode = rfc4733 device_state_busy_at = 2 force_rport = no rtp_symmetric = no rewrite_contact = no identify_by = username outbound_proxy = chi-sbc3-iad.bluip.com trust_id_outbound = yes from_domain = amtelco.com disallow = all allow = ulaw [identify72] type = identify endpoint = didassignedbybluip match = bluipaddress [registration2] type = registration transport = transport1 client_uri = sip:didassignedbybluip at companyname dot com server_uri = sip:chi-sbc3-iad.bluip.com...

I am trying to configure a connection to BluIP. I am able to make incoming calls work. However outgoing calls are not working. For the Outbound Registration, I noticed the contact field is always the internal IP address of my pc instead of mycompany dot com I can Originate (using AMI) to my Vitelity trunk (IP based authentication). However, when I Originate to my BluIP, it is being rejected.

...=asterisk auth=DEADDEADBEEF 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.co...

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)