[Feb 28 08:53:20] NOTICE[391515][C-0000001c]: chan_sip.c:26601 handle_request_invite: Failed to authenticate device

gjyotin305 · February 28, 2024, 8:54am

I am using asterisk 16, It is showing connected in my Softphone on my IOS device but i am not able to use my dialplan.

Please help

Pooh · February 28, 2024, 9:22am

On Wednesday 28 February 2024 at 10:05:36, gjyotin305 via Asterisk Community
wrote:

I am using asterisk 16, It is showing connected in my Softphone on my IOS
device but i am not able to use my dialplan.

Please help

For us to help you, you have to help us.

Show us (the relevant part of) your dialplan.
Explain in reasonable detail what you mean by “I am not able to use” it.
This sounds like a pretty beginner’s question, so why are you using
Asterisk 16 and chan_sip?

We have no idea what you have set up until you tell us. Until we know that,
we have no idea what the problem might be.

Antony.

–
I’m not impossible, just highly implausible.

                                               Please reply to the list;
                                                     please *don't* CC me.

gjyotin305 · February 28, 2024, 11:05am

Yeah I am trying to setup a simple asterisk server such that I can call it and there is a music playback for now.

[general]
context=internal
allowguest=no
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
disallow=all
allow=ulaw
alwaysauthreject=yes
canreinvite=no
nat=yes
session-timers=refuse
localnet=192.168.1.0/255.255.255.0
externip=*********

[7001]
type=friend
host=dynamic
secret=****
context=internal

[7002]
type=friend
host=dynamic
secret=****
context=internal

and extensions.conf

[internal]
exten => 701,1,Answer()
exten => 701,2,Playback(vm-nobodyavail)

gjyotin305 · February 28, 2024, 11:22am

Also if you can tell me where i look up some resources to use chan_pjsip it would be much appreciated because it’s very hard to find resources to learn asterisk.

gjyotin305 · February 28, 2024, 2:33pm

Here my SIP phone is connecting but i am unable to make the call to 701

david551 · February 28, 2024, 2:42pm

https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/res_pjsip-Configuration-Examples/

github.com

asterisk/asterisk/blob/master/configs/samples/pjsip.conf.sample

; PJSIP Configuration Samples and Quick Reference
;
; This file has several very basic configuration examples, to serve as a quick
; reference to jog your memory when you need to write up a new configuration.
; It is not intended to teach PJSIP configuration or serve as an exhaustive
; reference of options and potential scenarios.
;
; This file has two main sections.
; First, manually written examples to serve as a handy reference.
; Second, a list of all possible PJSIP config options by section. This is
; pulled from the XML config help. It only shows the synopsis for every item.
; If you want to see more detail please check the documentation sources
; mentioned at the top of this file.

; ============================================================================
; NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE NOTICE
;
; This file does not maintain the complete option documentation.
; ============================================================================

This file has been truncated. show original

https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships/

For the full details:

https://docs.asterisk.org/Configuration/Channel-Drivers/SIP/Configuring-res_pjsip/PJSIP-Configuration-Sections-and-Relationships/#config-section-help-and-defaults

david551 · February 28, 2024, 2:44pm

Please capture CLI output as plain text, and also provide the verbose full log.

FrancoSmash · February 28, 2024, 2:50pm

Try changing your bindport=5060 to 5070 & allow it tru your firewall
that should change your “unmonitored” and change qualify=yes

david551 · February 28, 2024, 3:22pm

unmonitored is not an error condition. It is setting qualify to yes that will make it monitored, not changing the port number. The only advantage of going to 5070 is that you might prevent the router recognizing SIP and mishandling it - it is better to disable such features in the router. It is not a big enough change to increase security . (Being unmonitored may mean that temporary router rules time out, but is a setting that can sometimes be valid and necessary.)

Given that you are successfully registering, the authentication error is probably the expected result of being open to the internet; you will get many many toll fraud attempts.

There is no valid reason for type=friend here, and, as it can reduce security (although probably not in this case) and cause other problems, it would be better to use type=peer.

gjyotin305 · February 29, 2024, 8:55am

Asterisk 16.2.1~dfsg-2ubuntu1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 16.2.1~dfsg-2ubuntu1 currently running on ubuntu-s-1vcpu-2gb-blr1-01 (pid = 411533)
ubuntu-s-1vcpu-2gb-blr1-01*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
7001                      (Unspecified)                            D  Yes        Yes            0        Unmonitored       
7002/7002                 *************                           D  Yes        Yes            35060    Unmonitored       
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 1 offline]
ubuntu-s-1vcpu-2gb-blr1-01*CLI>

I have connected the 7002 name and I am unable to dial 701 in the softphone
How do I get the verbose full log??

gjyotin305 · February 29, 2024, 8:59am

Asterisk 16.2.1~dfsg-2ubuntu1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 16.2.1~dfsg-2ubuntu1 currently running on ubuntu-s-1vcpu-2gb-blr1-01 (pid = 413740)
ubuntu-s-1vcpu-2gb-blr1-01*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
7001                      (Unspecified)                            D  Yes        Yes            0        UNKNOWN           
7002/7002                 ***********                         D  Yes        Yes            35060    OK (160 ms)       
2 sip peers [Monitored: 1 online, 1 offline Unmonitored: 0 online, 0 offline]
ubuntu-s-1vcpu-2gb-blr1-01*CLI>

After enabling qualify=yes

gjyotin305 · February 29, 2024, 9:00am

I dont understand why my softphone is unable to call is it because of an issue with IOS softphones, because it is clearly connecting with my server

gjyotin305 · February 29, 2024, 9:05am

Also what are some standard things which I can playback is there a list of these things.If yes, then where can I find it?

david551 · February 29, 2024, 11:24am

https://docs.asterisk.org/Operation/Logging/Collecting-Debug-Information/

system · March 30, 2024, 11:25am

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