similar to: Asterisk Can't start with the default configs

Hi, I tried to run the make progdocs, but the first time, it said, I have no doxygen installed. So I compiled the latest release and reconfigure the asterisk. And after it, ut sucessfully started to build the docs. But it took a lot of time, So finally I aborted the process... Is there a way to disable doc creating? The --disable-xmldoc is enough? Thanks! On Dec 2, 2016 3:36 PM, "Joshua

I am trying to launch Asterisk on a different directory with the parameter 'C asterisk -vvvvvvvvvvvvvvvvvvgc -C /etc/asterisk1/asterisk.conf Parsing '/etc/asterisk1/extconfig.conf': Found Resetting translation matrix UUID system initiated Parsing /etc/asterisk1/asterisk.conf == Parsing '/etc/asterisk1/asterisk.conf': Found Not changing threadpool size since new size 0 is

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under

Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is 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

Is there some documentation for all the available sorcery.conf mappings for realtime? Asterisk already includes some tables in the database that are not enabled by default on the sorcery.conf like transports and outbound registrations. There are no examples in the file on how to enable them. Where can I find some documentation to enable those mappings? -- Telecomunicaciones Abiertas de

Hello all, We would like to pull the RTT of registered endpoints from MySQL for use in a webportal. However it doesn't appear asterisk tables this by default like chan_sip did. I've found some information [1] that a modification of sorcery.conf can get it writing to a table. But I'm struggling to figure out what that configuration is. It seems the object is called

Matt Hoskins wrote: > Good Morning, > > After further investigation, I found that the res_pjsip_publish_asterisk > module does not use the realtime sorcery wizard, but instead only reads > from the configuration files. I've been able to patch the module, using > the logic from the other modules to learn how to make the sorery > configuration read from the other sorcery

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

I use realtime for my Asterisk configuration and are now making the transition to Asterisk 13 and PJSIP. I used alchemy to set up my databases and I can now configure my endpoints. While trying to configure a trunk I can see that there is a database table called ps_registrations that should be used to make the registration to the provider but there is no corresponding entry in the

Hi list. I use sorcery to configure an astdb backend to my pjsip endpoints. This works well, but it would be even better if I could set attribute defaults for these endpoints in the config file. The way I do it now forces me to store all endpoint attributes for each endpoint, even when most of them are effectively defaults. If I need to change one, I need to update each endpoint in the astdb.

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

On Thu, Sep 14, 2017, at 05:27 PM, Bryant Zimmerman wrote: > This appears to be some kind of cache issue. > We have been doing caching with earlier versions of asterisk 13 on the > pjsip realtime, but now for some reason > The items only show up the first time we use pjsip list/show and then > they > are wiped. I see a new full cache option and that appears to make a >

Howdy, Is there a way to use realtime with phoneprov.com and pjsip? I've got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision. I was hoping the Sorcery page in the wiki would help possibly but it's blank :( https://wiki.asterisk.org/wiki/display/AST/Sorcery -- A human being should be able to change a

Joshua We have completed more testing this morning and when we remove the realtime cache options from the sorcery file the endpoints complete registration, but we pjsip show/list does not offer any feed back at all, We also can't send any pjsip send notify commands as they say they don't have an endpoint there. Something has changed in the cache part of the system that is breaking

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

No, the disable-xmldoc doesn't disable the whole doc creating procedure. Is there a way to disable it completely? Regarding the issue... Of course, I Can open a ticket, just I don't know about what exactly. I want to compile it without doc generate to make the asterisk module loads up fine. On Dec 4, 2016 8:41 PM, "Joshua Colp" <jcolp at digium.com> wrote: > On Sun,

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,

On Fri, Sep 15, 2017, at 10:37 AM, Bryant Zimmerman wrote: > Joshua > > That is the interesting part of it. We took our configs and database > tables from our working 13.12.2 deployments and tried to use them with > our > new 13.17.1 deployments and we are having issues where the tables are not > working. On the new server asterisk keeps saying it can't find the

Hello configuration as follows, dial-out works: capi.conf: [hfcpci] ;;PointToPoint (55512-0) isdnmode=MSN incomingmsn=* ;msn=61 controller=1 devices=2 context=incoming extensions.conf: [incoming] exten => _XX,1,Playback(demo-abouttotry) exten => _XX,n,Dial,SIP/xlite1 exten => _XX,n,HangUp When call is placed, the following debug info is shown, after the last line, it stalls until

Hi! Installing new asterisk server and decided to use chan_pjsip. While module reload I get: y 12 15:33:04] ERROR[21137]: config_options.c:715 aco_process_var: Could not find option suitable for category '3567' named 'inband_progress' at line 867 of [May 12 15:33:04] ERROR[21137]: res_sorcery_config.c:317 sorcery_config_internal_load: Could not create an object of type