TIPS pjsip.conf could not be loaded

troysmoke

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Good Morning,

I just rebuilt my Raspberry Pi with the latest Incredible PBX build (13.13.10). I have successfully added my extensions and set up Skyetel trunks, routes, etc.

However, when I attempt to make a call, the call fails immediately.

The asterisk log files show most errors on reboot:
[2019-05-16 08:19:31] ERROR[1889] ari/config.c: No configured users for ARI
[2019-05-16 08:19:31] ERROR[1889] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1:
[2019-05-16 08:19:32] ERROR[1957] res_pjsip.c: Unable to register module idle monitor module
[2019-05-16 08:19:32] ERROR[1889] pbx_functions.c: Function ODBC_TIMECLOCK already registered.
[2019-05-16 08:19:32] ERROR[1889] pbx_functions.c: Function ODBC_ASTERIDEX already registered.
[2019-05-16 08:19:33] ERROR[1889] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory

These errors repeat a lot:
[2019-05-16 09:06:31] ERROR[6541] res_sorcery_config.c: Config file 'pjsip.conf' could not be loaded; configuration contains a duplicate object: 'global' of type 'global'
[2019-05-16 09:06:31] ERROR[6540] ari/config.c: No configured users for ARI

Is this something fixable? I have already rebuilt the image once.

Thanks for your help!
troysmoke
 

troysmoke

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Nevermind...config error in the outbound route - I put in the CID in the dial manipulation rules...

I still have the no audio issue, but that is a different thread.
 

l4cky

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Good Morning,

I just rebuilt my Raspberry Pi with the latest Incredible PBX build (13.13.10). I have successfully added my extensions and set up Skyetel trunks, routes, etc.

However, when I attempt to make a call, the call fails immediately.

The asterisk log files show most errors on reboot:
[2019-05-16 08:19:31] ERROR[1889] ari/config.c: No configured users for ARI
[2019-05-16 08:19:31] ERROR[1889] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1:
[2019-05-16 08:19:32] ERROR[1957] res_pjsip.c: Unable to register module idle monitor module
[2019-05-16 08:19:32] ERROR[1889] pbx_functions.c: Function ODBC_TIMECLOCK already registered.
[2019-05-16 08:19:32] ERROR[1889] pbx_functions.c: Function ODBC_ASTERIDEX already registered.
[2019-05-16 08:19:33] ERROR[1889] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory

These errors repeat a lot:
[2019-05-16 09:06:31] ERROR[6541] res_sorcery_config.c: Config file 'pjsip.conf' could not be loaded; configuration contains a duplicate object: 'global' of type 'global'
[2019-05-16 09:06:31] ERROR[6540] ari/config.c: No configured users for ARI

Is this something fixable? I have already rebuilt the image once.

Thanks for your help!
troysmoke


same problem with me, can u tell me how u fixed it?
 

troysmoke

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Never fixed it, but it looks like the errors don't interfere with normal operation. My PBX is up and running with Skyetel trunks and Yealink phones. I'm using chan_sip.

Someone who actually knows something about Asterisk will have to chime in on why those errors might be showing up.
 

kenn10

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PJSIP is a minefield best left alone. Despite all the hype, it is not ready for prime time. If you stick with chan_sip, you won't have as many issues.

Make sure in the Settings > Asterisk SIP Settings > Chan_SIP Settings that the bind port is set to 5060 and in the PJSIP settings, that the port is set to 5061. Then set up your trunks and extensions using Chan_SIP and port 5060. You'll save yourself a lot of pain.
 

l4cky

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There is an instruction with google voice and obihai that ask for pjsip but no pjsip works.. because of these errors..
 

billsimon

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PJSIP is a minefield best left alone. Despite all the hype, it is not ready for prime time. If you stick with chan_sip, you won't have as many issues.

Please refrain from spreading FUD.

If it doesn't work, it is probably because you are setting it up wrong.

Granted, the configuration of pjsip is more complicated and that makes it more difficult to set up. But that does not make it "a minefield" nor does it make it "not ready for prime time."
 

billsimon

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My point stands... it may be challenging but it is not "not ready for primetime" and statements like that just keep people away from it because they assume it's broken/buggy. On the other hand, if more people dug into it and learned it, then we could all move ahead faster.

kenn10 also labeled himself as "Guru-ish" - so frankly I'm gonna expect more :)
 

dallas

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My point stands... it may be challenging but it is not "not ready for primetime" and statements like that just keep people away from it because they assume it's broken/buggy. On the other hand, if more people dug into it and learned it, then we could all move ahead faster.

So how about a few pointers (and gotyas to avoid) on getting it going as it will be a long time before pjsip replaces chan_sip, and that's because it just works.
 
Last edited:

Tonyclewis

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PJSIP is a minefield best left alone. Despite all the hype, it is not ready for prime time. If you stick with chan_sip, you won't have as many issues.

Make sure in the Settings > Asterisk SIP Settings > Chan_SIP Settings that the bind port is set to 5060 and in the PJSIP settings, that the port is set to 5061. Then set up your trunks and extensions using Chan_SIP and port 5060. You'll save yourself a lot of pain.
This is a very dangerous and broad statement. The Asterisk team no longer supports Chan SIP and only PJSIP. If someone is having a specific issue we should work to educate or assist them not tell them to use a deprecated piece of software that won’t even get patched anymore.
 

krzykat

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Hello @Tonyclewis , Why is Chan SIP going to be deprecated? I've been using it for quite some time and haven't seen the need to use PJsip. When I tried PJsip when it first came out, I experienced some issues with it, but I guess you are saying that all those issues should be resolved?
 

l4cky

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Can you post your pjsip.conf file for us to look at.

Not sure which ones, but I copy paste them all here. They should be the default files after installation...?

pjsip.conf (ASTERISK SYSTEM CONFIGURATION FILES)

Code:
;--------------------------------------------------------------------------------;
;          Do NOT edit this file as it is auto-generated by FreePBX.             ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate.                                                         ;
;--------------------------------------------------------------------------------;
#include pjsip_custom.conf
#include pjsip.transports.conf
#include pjsip.transports_custom_post.conf
#include pjsip.endpoint.conf
#include pjsip.endpoint_custom_post.conf
#include pjsip.aor.conf
#include pjsip.aor_custom_post.conf
#include pjsip.auth.conf
#include pjsip.auth_custom_post.conf
#include pjsip.registration.conf
#include pjsip.registration_custom_post.conf
#include pjsip.identify.conf
#include pjsip.identify_custom_post.conf


[global]
type=global
user_agent=FPBX-13.0.195.28(13.22.0)
#include pjsip_custom_post.conf

pjsip_custom.conf (ASTERISK CUSTOM CONFIGURATION FILES)

Code:
; pjsip_custom, Copyright (c) 2018, Ward Mundy & Associates, LLC.
; Licensed for use with NAF's Asterisk 13 PJSIP patches pursuant to GPL2.

[incoming-registrations-unused-but-required]
type=transport
protocol=udp
bind=0.0.0.0:5065

[global]
type=global
debug=true
keep_alive_interval=90

[transport_tls]
type=transport
protocol=tls
bind=0.0.0.0:5062


Then the Error is (in Asterisk log):

[2019-06-07 03:25:30] ERROR[27862] res_sorcery_config.c: Config file 'pjsip.conf' could not be loaded; configuration contains a duplicate object: 'global' of type 'global'
[2019-06-07 03:25:30] ERROR[27861] ari/config.c: No configured users for ARI
 

wardmundy

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Comment out the [global] context in pjsip_custom.conf, and you should be good to go:
Code:
;[global]
;type=global
;debug=true
;keep_alive_interval=90
 

l4cky

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Comment out the [global] context in pjsip_custom.conf, and you should be good to go:
Code:
;[global]
;type=global
;debug=true
;keep_alive_interval=90


Thanks Ward for the clear instructions, that solves the problem but when I restart asterisk. It has those errors:

Code:
[2019-06-07 18:09:13] ERROR[2018] ari/config.c: No configured users for ARI
[2019-06-07 18:09:13] ERROR[2018] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1:
[2019-06-07 18:09:13] ERROR[13756] res_pjsip.c: Unable to register module idle monitor module
[2019-06-07 18:09:13] ERROR[2018] pbx_functions.c: Function ODBC_TIMECLOCK already registered.
[2019-06-07 18:09:13] ERROR[2018] pbx_functions.c: Function ODBC_ASTERIDEX already registered.
[2019-06-07 18:09:14] ERROR[2018] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory
[2019-06-07 18:12:57] Asterisk 13.22.0 built by root @ incrediblepbx on a armv7l running Linux on 2018-08-25 13:33:46 UTC
[2019-06-07 18:12:58] ERROR[14329] ari/config.c: No configured users for ARI

what should I do?
 

l4cky

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Nothing. Everything you need is working.

When I try to add a pjsip ext 444, it doesn't work with these errors and i can't registrer the ext 444 to a softphone.

[2019-06-19 13:44:55] ERROR[14498] res_pjsip_endpoint_identifier_ip.c: Identify '444-identify' is not configured to match anything.
[2019-06-19 13:44:55] ERROR[14498] res_sorcery_config.c: Could not create an object of type 'identify' with id '444-identify' from configuration file 'pjsip.conf'
 

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