I installed ASTPP but not able to receive DID calls on any destination

General questions DID
1

2023-04-16 20:20:08.083038 99.03% [WARNING] sofia_reg.c:1846 SIP auth challenge (INVITE) on sofia profile ‘default’ for [01146812410595@167.99.155.83] from ip 132.148.78.229

2023-04-16 20:20:08.083038 99.03% [DEBUG] switch_core_state_machine.c:600 (sofia/default/1000@167.99.155.83) State NEW

2023-04-16 20:20:08.083038 99.03% [DEBUG] sofia.c:2419 detaching session 943b72ae-201b-4405-b22f-a80b03e817e1

2023-04-16 20:20:08.203041 99.03% [DEBUG] sofia.c:2532 Re-attaching to session 943b72ae-201b-4405-b22f-a80b03e817e1

2023-04-16 20:20:08.223044 99.03% [INFO] sofia.c:10453 sofia/default/1000@167.99.155.83 receiving invite from 132.148.78.229:50010 version: 1.10.9 -release-21-a615e85afc 64bit call-id: 1476197229-1595040756-720665475

2023-04-16 20:20:08.223044 99.03% [DEBUG] sofia.c:10547 verifying acl “default” for ip/port 132.148.78.229:0.

2023-04-16 20:20:08.223044 99.03% [WARNING] sofia_reg.c:3211 Can’t find user [1000@167.99.155.83] from 132.148.78.229

You must define a domain called ‘167.99.155.83’ in your directory and add a user with the id=“1000” attribute

and you must configure your device to use the proper domain in its authentication credentials.

2023-04-16 20:20:08.223044 99.03% [WARNING] sofia_reg.c:1791 SIP auth failure (INVITE) on sofia profile ‘default’ for [01146812410595@167.99.155.83] from ip 132.148.78.229

2

Hello @Steven ,
Your logs are unclear to us. Could you describe the call flow in more detail?
The DID number is dialed from where?

3

I Have Created Below Mention details :-

Customer
Provider
SIP Device
IP Setting
Rate Group
Origination Rate
gateway
trunks
termination rate
DID

And Assigned DID to External PBX IP. When I’m dialing from other numbers on that particular IP one beep coming and call disconnected.

4

My Customer and Provider already syncs

freeswitch@markettter> sofia status
Name Type Data State

              default	profile	          sip:mod_sofia@167.99.155.83:5060	RUNNING (0)
 default::EV1_TELECOM	gateway	                 sip:FreeSWITCH@45.63.5.73	NOREG

=================================================================================================
1 profile 0 aliases

recv 544 bytes from udp/[5.9.100.170]:5060 at 05:54:28.364361:

OPTIONS sip:markettter.shop SIP/2.0
Via: SIP/2.0/UDP 5.9.100.170:5060;branch=z9hG4bK0dc132bf;rport
Max-Forwards: 70
From: “asterisk” sip:asterisk@5.9.100.170;tag=as53c449c9
To: sip:markettter.shop
Contact: sip:asterisk@5.9.100.170:5060
Call-ID: 43e1b6172d2746f20de1619c064180a0@5.9.100.170:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.25.1-vici
Date: Mon, 17 Apr 2023 05:54:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

send 672 bytes to udp/[5.9.100.170]:5060 at 05:54:28.364619:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 5.9.100.170:5060;branch=z9hG4bK0dc132bf;rport=5060
From: “asterisk” sip:asterisk@5.9.100.170;tag=as53c449c9
To: sip:markettter.shop;tag=Kva72r5Uv3K2Q
Call-ID: 43e1b6172d2746f20de1619c064180a0@5.9.100.170:5060
CSeq: 102 OPTIONS
Contact: sip:167.99.155.83
User-Agent: ASTPP
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

5

Is there anyone who can help me on this ?

6

I Have Created Below Mention details :-

Customer
Provider
SIP Device
IP Setting
Rate Group
Origination Rate
gateway
trunks
termination rate
DID

And Assigned DID to External PBX IP. When I’m dialing from other numbers on that particular IP one beep coming and call disconnected.

7

Hello @Steven ,
Please take the logs in Pastebin and share it here so I can check accordingly.
https://pastebin.freeswitch.org/

8

https://pastebin.freeswitch.org/view/b909d6c7

9

While doing outbound calls getting error INCOMING_CALL_BARRED [54]

SIP Profile Settings :

rtp-ip 164.90.182.64
dialplan XML
debug 0
sip-trace no
tls FALSE
inbound-reg-force-matching-username TRUE
disable-transcoding TRUE
all-reg-options-ping FALSE
unregister-on-options-fail TRUE
log-auth-failures TRUE
status 0
inbound-bypass-media FALSE
inbound-proxy-media FALSE
disable-transfer FALSE
enable-100rel FALSE
rtp-timeout-sec 300
dtmf-duration 2000
manual-redirect TRUE
aggressive-nat-detection TRUE
enable-timer FALSE
minimum-session-expires 120
session-timeout-pt 1800
auth-calls TRUE
apply-inbound-acl default
inbound-codec-prefs PCMA,PCMU,g729
outbound-codec-prefs PCMA,PCMU,g729
inbound-late-negotiation FALSE
sip-capture no
forward-unsolicited-mwi-notify FALSE
context default
rfc2833-pt 101
rtp-timer-name soft
hold-music $${hold_music}
manage-presence TRUE
presence-hosts $${domain},$${local_ip_v4}
presence-privacy $${presence_privacy}
inbound-codec-negotiation generous
auth-all-packets FALSE
ext-rtp-ip 164.90.182.64
ext-sip-ip 164.90.182.64
rtp-hold-timeout-sec 1800
force-register-domain $${domain}
force-subscription-domain $${domain}
force-register-db-domain $${domain}
challenge-realm auto_from
nonce-ttl 60
pass-callee-id FALSE
dtmf_type rfc2833
10

Hi @Steven ,
We have checked the logs and it seems the gateway is rejecting the calls. Please check once with your gateway provider.