search for: ps_endpoints

Hello, In Asterisk 11 it is possible to set extensions on DB table (sipppers) and also in sip.conf. But in Asterisk 13 apparently this is not possible: as I tried to set in ps_endpoints and also in pjsip.conf but only the realtime endpoints are loaded. Is there a way to use realtime + static modes at the same time for the ps_endpoints lookup using PJSIP. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/a...

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Annus Fictus Sent: 13 June 2016 14:11 To...

Hello, I wonder if anybody is using PJSIP realtime in production environment? I've started to play with it and encountered many problems. Here's my config: sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints extconfig.conf: [settings] ps_endpoints => pgsql,users,pjsip_endpoints_v pjsip_endpoints_v is postgresql view. 1. The biggest problem: if I have small number of endpoints (roughly up to a 100) then asterisk loads ok and pjsip seems to be working ok (with other problems described below). If I...

Sunny wrote: > Hello, > > > In Asterisk 11 it is possible to set extensions on DB table (sipppers) > and also in sip.conf. > > But in Asterisk 13 apparently this is not possible: as I tried to set in > ps_endpoints and also in pjsip.conf but only the realtime endpoints are > loaded. > > Is there a way to use realtime + static modes at the same time for the > ps_endpoints lookup using PJSIP. Yes. The sorcery.conf file controls this and allows specifying multiple things. It would be like follows:...

...the ip addresses? Is there any practical use for such a setup? Also I notice, an AOR does seem do be directly correlated with an auth record, so why are they separate in the configuration, why not unify the aor and the auth objects? And, while I'm at it, in the realtime tables, the length of ps_endpoints.aors = 200, and the length of ps_endpoints.auths = 40. This suggests there are scenarios where there are aors, without corresponding auth. Can you mix dynamic and static AORs within one endpoint, and would there be a use case for that? Thanks, Antonio. > > >> So why does aor have...

.......................> <Hash....> <Status> <RTT(ms)..> ========================================================================================= Contact: pbx-node-1/sip:myurl.ch:5060 771bf6a7d4 Avail 8.833 The extconfig.conf file looks like this: [settings] ps_endpoints => odbc,asterisk ps_auths => odbc,asterisk ps_aors => odbc,asterisk ps_domain_aliases => odbc,asterisk ps_endpoint_id_ips => odbc,asterisk ps_contacts => odbc,asterisk extensions => odbc,asterisk Any idea why I need to reload PJSIP if I want the aors to be qualified? Cheers...

...jcolp at digium.com> wrote: > Sunny wrote: > >> Hello, >> >> >> In Asterisk 11 it is possible to set extensions on DB table (sipppers) >> and also in sip.conf. >> >> But in Asterisk 13 apparently this is not possible: as I tried to set in >> ps_endpoints and also in pjsip.conf but only the realtime endpoints are >> loaded. >> >> Is there a way to use realtime + static modes at the same time for the >> ps_endpoints lookup using PJSIP. >> > > Yes. The sorcery.conf file controls this and allows specifying multiple &...

Hello, Is it possible to mix PJSIP realtime and flat file configuration in pjsip,conf? What we want is to set up endpoints in the ps_endpoints table with some columns set but most being NULL, and then allow end-customers to optionally add configuration by adding a pjsip.conf section. For example, in ps_endpoinds might be an endpoint with id "asterisk-1" with the transport, aors, auth, and context columns set but all other field...

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. Any pointers? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +...

...only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't have a possible misconfiguration issue. Here is my sorcery.conf and extconfig file: Sorcery.conf ---------------- [res_pjsip] endpoint=realtime,ps_endpoints auth=realtime,ps_auths aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips [res_pjsip_outbound_publish] outbound-publish=realtime,ps_outbound_publish [res_pjsip_pubsub] inbound-publication=realtime,ps_inbound_public...

...objects? They aren't at all. Auth = Authentication. Used to authenticate incoming calls/registrations/other stuff, or used to authenticate outgoing things. They are NOT the same. AOR is a name for reaching something. > And, while I'm at it, in the realtime tables, the length of > ps_endpoints.aors = 200, and > the length of ps_endpoints.auths = 40. This suggests there are scenarios > where there > are aors, without corresponding auth. Can you mix dynamic and static > AORs within one > endpoint, and would there be a use case for that? You can mix however you want. -- Jo...

...only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't have a possible misconfiguration issue. Here is my sorcery.conf and extconfig file: Sorcery.conf ---------------- [res_pjsip] endpoint=realtime,ps_endpoints auth=realtime,ps_auths aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips [res_pjsip_outbound_publish] outbound-publish=realtime,ps_outbound_publish [res_pjsip_pubsub] inbound-publication=realtime,ps_inbound_public...

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Hello, I am moving from realtime chan_sip to pjsip and one of the problem I am facing is the lack of "sipregs". With chan_sip, when an extension registers, the server where it has registered to is stored in sipregs. Is there something similar in pjsip? How can I find on which server the pjsip extension has registered to? Leandro -------------- next part -------------- An HTML

Hello, I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact. I am having trouble mapping them to the typical SIP configuration settings on a phone. Suppose I have a phone with two line buttons, for two extension numbers. Now, I think that means two 'endpoints' in pjsip right? But what exactly is the difference between

...shua Colp) * ASTERISK-20858 - app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) * ASTERISK-16777 - several filename bugs in Record() application (Reported by klaus3000) * ASTERISK-27168 - alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) * ASTERISK-19103 - When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function ha...

...ASTERISK-16777 - several filename bugs in Record() application (Reported by klaus3000) * ASTERISK-27209 - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used (Reported by Torrey Searle) * ASTERISK-27168 - alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-19103 - When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used. (Reported by Jim Van...

...s Terhune) * ASTERISK-20858 - app_minivm fails to clean up mkstemp files (Reported by Walter Doekes) * ASTERISK-16777 - several filename bugs in Record() application (Reported by klaus3000) * ASTERISK-27168 - alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-23608 - ControlPlayback fails to play files with names containing certain non-alpha characters (Reported by Jonathan White) * ASTERISK-19103 - When using realtime queues, function QUEUE_MEMBER_LIST() will return a...

...els: channel_state_invalid always leaks snapshot reference. (Reported by Marin Odrljin) * ASTERISK-27379 - stream: Allow streams on a topology to be put into groups (Reported by Joshua Colp) * ASTERISK-27374 - alembic: PJSIP scripts are missing column bundle in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' (Reported by Anthony Mes...

...els: channel_state_invalid always leaks snapshot reference. (Reported by Marin Odrljin) * ASTERISK-27379 - stream: Allow streams on a topology to be put into groups (Reported by Joshua Colp) * ASTERISK-27374 - alembic: PJSIP scripts are missing column bundle in ps_endpoints table (Reported by Florian Floimair) * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage documentation (Reported by Igor Goncharovsky) * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function 'imap_delete_old_greeting' (Reported by Anthony Mes...