search for: allow_subscrib

I am still a little confused about how to activate MWI with PJSIP on Asterisk 13.9.1. I use realtime for configuration. So far I have tried setting both the mailboxes field on ps_endpoints and the mailboxes field in ps_aors but I cannot get the indicator lamp to blink on any of my phones (Digium, Aastra and Yealink). I have tried just the number of the mailbox and also adding the context.

...Ext11 - LocA/Ext11's CallerID displays "10", but LocA/Ext10 calling LocB/Ext20 displays "Location A - 10" for caller ID. I have my locations set up in the pjsip_wizard.conf file like this: Location A: [Uplink_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/allow = !all,g729 aor/qualify_frequency = 30 registration/expiration = 1800 [Location B](Uplink_defaults) endpoint/context = internal remote_hosts = 10.10.20.253:5060 sends_registrations = no accepts_registrations = no sends_auth = no accepts_auth = no Location B: [Uplink_defaults...

...pe=auth username = Biv_Sortie password=3314C9BA9688C2AA [Biv_Sortie] type=endpoint context = Isdn_Inbound dtmf_mode=rfc4733 disallow=all allow = g722, alaw, g729 direct_media=no trust_id_inbound = yes send_rpid=yes from_user = 75B55BTQUHSG at orange-obs.fr from_domain = orange-obs.fr language = en allow_subscribe = yes auth = Biv_Exit outbound_auth = Biv_Sortie aors = Biv_Sortie Question how can I solve this character problem "@"? 2) resolution of the orange-obs.fr DNS.  I am attaching an extract from the documentation that Orange issued in 2015 SIP/Internet is described in RFC3261 and follow...

...s We are using the wizard to configure our pjsip trunk(see below) How do we get this setting to work contact_user=username We want to change the contact field in the sip invite to display the username of the trunk [trunk_defaults](!) type = wizard transport = transport-udp endpoint/allow_subscribe = no endpoint/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...

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:

...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 subscribe message and reply as recored...

...t] type=transport protocol=udp bind=0.0.0.0:5160 local_net=192.0.0.0/24 external_media_address=192.168.0.196 external_signaling_address=192.168.0.196 [6001] type=endpoint context=callcenter disallow=all allow=g729 allow=ulaw allow=gsm auth=6001 aors=6001 transport=transport-udp-nat direct_media=no allow_subscribe=yes sub_min_expiry=30 [6001] type=auth auth_type=userpass password=6001 username=6001 [6001] type=aor max_contacts=99 Has anyone experienced the same problem? I upgraded my asterisk to 17 and the problem still persists. Thanks! -------------- next part -------------- An HTML attachment was...

Hi Joshua thank you for the quick reply > Have you checked the Asterisk console when PJSIP is loaded to see if > the endpoint did not load for some reason? Does it show up in "pjsip > show endpoints"? Yes, the endpoint shows up. Endpoint: 11/(scrubbed from mail) Not in use 0 of inf InAuth: 11/11 Aor: 11

...p=false record_off_feature=automixmon send_diversion=true language= mwi_from_user= rtp_ipv6=false ice_support=false callerid=unknown aggregate_mwi=true one_touch_recording=false cos_video=0 accountcode= allow=(g722|ulaw|alaw) rewrite_contact=false t38_udptl_ipv6=false tone_zone= user_eq_phone=false allow_subscribe=true rtp_engine=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...

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)