Asterisk 16 NOTICE

[jcolp]

I have increased the maximum number of allowed replies for a new user account in the first day.

[ABHASHbhai17]

thanks

[ABHASHbhai17]

[Jun 22 10:25:45] WARNING[8603]: res_config_pgsql.c:1443 parse_config: PostgreSQL RealTime: Not connected
[Jun 22 10:25:45] NOTICE[8603]: res_config_ldap.c:1832 parse_config: No directory user found, anonymous binding as default.
[Jun 22 10:25:45] ERROR[8603]: res_config_ldap.c:1858 parse_config: No directory URL or host found.
[Jun 22 10:25:45] NOTICE[8603]: res_config_ldap.c:1776 reload: Cannot reload LDAP RealTime driver.
[Jun 22 10:25:45] NOTICE[8603]: cdr.c:4585 cdr_toggle_runtime_options: CDR simple logging enabled.
[Jun 22 10:25:45] ERROR[8595]: res_pjsip/config_transport.c:736 transport_apply: Transport ‘transport-udp’ could not be started: Address already in use
[Jun 22 10:25:45] ERROR[8595]: res_sorcery_config.c:422 sorcery_config_internal_load: Could not create an object of type ‘transport’ with id ‘transport-udp’ from configuration file ‘pjsip.conf’
[Jun 22 10:25:45] NOTICE[8595]: sorcery.c:1348 sorcery_object_load: Type ‘system’ is not reloadable, maintaining previous values
[Jun 22 10:25:45] ERROR[8595]: res_sorcery_config.c:422 sorcery_config_internal_load: Could not create an object of type ‘registration’ with id ‘MONKEY_TRUNK’ from configuration file ‘pjsip.conf’
[Jun 22 10:25:45] WARNING[8603]: res_phoneprov.c:1233 get_defaults: Unable to find a valid server address or name.
[Jun 22 10:25:45] NOTICE[8603]: chan_skinny.c:8482 config_load: Configuring skinny from skinny.conf
[Jun 22 10:25:45] ERROR[8603]: res_sorcery_config.c:422 sorcery_config_internal_load: Could not create an object of type ‘registration’ with id ‘MONKEY_TRUNK’ from configuration file ‘pjsip.conf’
[Jun 22 10:25:45] NOTICE[8603]: cel_custom.c:92 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Jun 22 10:25:45] WARNING[8603]: pbx.c:8797 ast_context_verify_includes: Context ‘local’ tries to include nonexistent context ‘iaxtel700’
[Jun 22 10:25:45] WARNING[8603]: pbx.c:8797 ast_context_verify_includes: Context ‘local’ tries to include nonexistent context ‘iaxtel700’
[Jun 22 10:25:45] NOTICE[8603]: app_queue.c:9449 reload_queue_rules: queuerules.conf has not changed since it was last loaded. Not taking any action.

can someone tell why is this coming and what i need to provide you more

[ABHASHbhai17]

what i need to do in pjsip if my provider is using magnus billing

[Pooh]

It looks like you have taken a default set of configuration files and tried to
use them without adapting them for what you want (and more importantly, do not
want) to do with your system.

Antony.

[ABHASHbhai17]

any suggestion what i can do

[Pooh]

I can’t help feeling, after seeing all the questions you have raised today,
that it would be useful for you to tell us:

1. What are you actually trying to achieve (no great detail, just a good
   overview of what this thing you want to build is supposed to do)?

2. How much experience of working with Asterisk do you have (simple dialplans,
   pjsip configuration, AMI, ARI, anything you’ve worked with and feel comfortable
   doing)?

3. How well do you know the SIP protocol and how user agents (clients and
   servers) communicate with each other?

4. From question 1, how much of what you’re trying to build have you managed
   to get working, and how much are you struggling with?

I’ll also add: the more you can demonstrate what you’ve done already, and the
more you can provide relevant log files of where something is not working as
intended, and the more specific you can be about asking for help with some
specific part of what you’re trying to do, the more likely it is that people
can / will help.

This is not a commercial support service for creating your Asterisk dialplans.

This is a community of volunteers (some of whom also contribute to Asterisk
development) who ask for help when they’re stuck with something, and provide
help to others when they can.

Nearly all of us expect the people asking for help to have shown that they’ve
put in a reasonable amount of effort of their own, and learned about the tool
they’re trying to use, before coming here and asking for more detailed
assistance on something they don’t quite get.

Questions like the subject of one of your topics “Asterisk 16 Full
Configuration For Outbound calls with ARI” are unlikely to get much helpful
response in my opinion, because:

a) you’re asking about an unsupported version of Asterisk (16)

b) you don’t say why you want to do this with ARI and what else you’ve tried
instead

c) you don’t give any indication that you’ve learned about ARI and made some
progress of your own

d) you don’t even give a sufficient definition of what you want to happen.
“outbound calls” are all very well, but what are they supposed to connect to?
What’s at the initiating end? An outbound call starts somewhere - what is it?

e) the phrase “I also need help with config before ARI or AMI” means nothing to
us without more explanation.

I hope this helps you to get a better understanding of how to ask questions on
a forum such as this, which are most likely to give you the help you want.

Regards,

Antony.

[dsiemens]

Hi

The 1st recommendation is not to directly connect asterisk to the internet. It costs more money and time but proper isolation will greatly reduce attack foot prints. The industry term is session border controller on all points of ingress and egress into the pbx.

Many will use fail2ban to monitor logs for authentication attempts and then block IP’s based on failed attempts.

But if you have weak passcodes or don’t mitigate the attack surface its only a matter of time to be comprised. Good security is usually applying layers to to the stack to mitigate different attacks vectors.